From 719a4917404307073f6de427383886cad0612b14 Mon Sep 17 00:00:00 2001 From: reuk Date: Wed, 15 Apr 2020 18:04:06 +0100 Subject: [PATCH] Android: Use built-in Oboe --- .../DemoRunner/JuceLibraryCode/AppConfig.h | 324 +++++++++ .../JuceLibraryCode/AppConfig.h | 297 +++++++++ .../jucer_ProjectExport_Android.h | 123 ++-- .../juce_audio_devices/juce_audio_devices.cpp | 6 +- .../juce_audio_devices/juce_audio_devices.h | 12 +- .../native/juce_android_Oboe.cpp | 20 +- .../native/oboe/CMakeLists.txt | 91 +++ .../juce_audio_devices/native/oboe/LICENSE | 202 ++++++ .../native/oboe/include/oboe/AudioStream.h | 523 +++++++++++++++ .../oboe/include/oboe/AudioStreamBase.h | 201 ++++++ .../oboe/include/oboe/AudioStreamBuilder.h | 424 ++++++++++++ .../oboe/include/oboe/AudioStreamCallback.h | 123 ++++ .../native/oboe/include/oboe/Definitions.h | 500 ++++++++++++++ .../native/oboe/include/oboe/LatencyTuner.h | 118 ++++ .../native/oboe/include/oboe/Oboe.h | 37 ++ .../oboe/include/oboe/ResultWithValue.h | 155 +++++ .../oboe/include/oboe/StabilizedCallback.h | 75 +++ .../native/oboe/include/oboe/Utilities.h | 73 +++ .../native/oboe/include/oboe/Version.h | 92 +++ .../juce_audio_devices/native/oboe/readme.md | 10 + .../native/oboe/src/aaudio/AAudioLoader.cpp | 348 ++++++++++ .../native/oboe/src/aaudio/AAudioLoader.h | 229 +++++++ .../oboe/src/aaudio/AudioStreamAAudio.cpp | 620 ++++++++++++++++++ .../oboe/src/aaudio/AudioStreamAAudio.h | 123 ++++ .../native/oboe/src/common/AudioClock.h | 75 +++ .../oboe/src/common/AudioSourceCaller.cpp | 38 ++ .../oboe/src/common/AudioSourceCaller.h | 83 +++ .../native/oboe/src/common/AudioStream.cpp | 211 ++++++ .../oboe/src/common/AudioStreamBuilder.cpp | 187 ++++++ .../src/common/DataConversionFlowGraph.cpp | 210 ++++++ .../oboe/src/common/DataConversionFlowGraph.h | 84 +++ .../oboe/src/common/FilterAudioStream.cpp | 72 ++ .../oboe/src/common/FilterAudioStream.h | 204 ++++++ .../oboe/src/common/FixedBlockAdapter.cpp | 38 ++ .../oboe/src/common/FixedBlockAdapter.h | 66 ++ .../oboe/src/common/FixedBlockReader.cpp | 73 +++ .../native/oboe/src/common/FixedBlockReader.h | 60 ++ .../oboe/src/common/FixedBlockWriter.cpp | 73 +++ .../native/oboe/src/common/FixedBlockWriter.h | 54 ++ .../native/oboe/src/common/LatencyTuner.cpp | 102 +++ .../native/oboe/src/common/MonotonicCounter.h | 112 ++++ .../native/oboe/src/common/OboeDebug.h | 46 ++ .../native/oboe/src/common/QuirksManager.cpp | 70 ++ .../native/oboe/src/common/QuirksManager.h | 52 ++ .../oboe/src/common/SourceFloatCaller.cpp | 30 + .../oboe/src/common/SourceFloatCaller.h | 44 ++ .../oboe/src/common/SourceI16Caller.cpp | 47 ++ .../native/oboe/src/common/SourceI16Caller.h | 48 ++ .../oboe/src/common/StabilizedCallback.cpp | 112 ++++ .../native/oboe/src/common/Trace.cpp | 75 +++ .../native/oboe/src/common/Trace.h | 31 + .../native/oboe/src/common/Utilities.cpp | 283 ++++++++ .../native/oboe/src/common/Version.cpp | 28 + .../native/oboe/src/fifo/FifoBuffer.cpp | 186 ++++++ .../native/oboe/src/fifo/FifoBuffer.h | 99 +++ .../native/oboe/src/fifo/FifoController.cpp | 31 + .../native/oboe/src/fifo/FifoController.h | 61 ++ .../oboe/src/fifo/FifoControllerBase.cpp | 71 ++ .../native/oboe/src/fifo/FifoControllerBase.h | 93 +++ .../oboe/src/fifo/FifoControllerIndirect.cpp | 31 + .../oboe/src/fifo/FifoControllerIndirect.h | 64 ++ .../native/oboe/src/flowgraph/ClipToRange.cpp | 38 ++ .../native/oboe/src/flowgraph/ClipToRange.h | 68 ++ .../oboe/src/flowgraph/FlowGraphNode.cpp | 111 ++++ .../native/oboe/src/flowgraph/FlowGraphNode.h | 422 ++++++++++++ .../src/flowgraph/ManyToMultiConverter.cpp | 47 ++ .../oboe/src/flowgraph/ManyToMultiConverter.h | 49 ++ .../src/flowgraph/MonoToMultiConverter.cpp | 43 ++ .../oboe/src/flowgraph/MonoToMultiConverter.h | 49 ++ .../native/oboe/src/flowgraph/RampLinear.cpp | 77 +++ .../native/oboe/src/flowgraph/RampLinear.h | 96 +++ .../src/flowgraph/SampleRateConverter.cpp | 64 ++ .../oboe/src/flowgraph/SampleRateConverter.h | 56 ++ .../native/oboe/src/flowgraph/SinkFloat.cpp | 50 ++ .../native/oboe/src/flowgraph/SinkFloat.h | 44 ++ .../native/oboe/src/flowgraph/SinkI16.cpp | 58 ++ .../native/oboe/src/flowgraph/SinkI16.h | 43 ++ .../native/oboe/src/flowgraph/SinkI24.cpp | 67 ++ .../native/oboe/src/flowgraph/SinkI24.h | 44 ++ .../native/oboe/src/flowgraph/SourceFloat.cpp | 43 ++ .../native/oboe/src/flowgraph/SourceFloat.h | 43 ++ .../native/oboe/src/flowgraph/SourceI16.cpp | 54 ++ .../native/oboe/src/flowgraph/SourceI16.h | 42 ++ .../native/oboe/src/flowgraph/SourceI24.cpp | 65 ++ .../native/oboe/src/flowgraph/SourceI24.h | 43 ++ .../resampler/HyperbolicCosineWindow.h | 68 ++ .../src/flowgraph/resampler/IntegerRatio.cpp | 52 ++ .../src/flowgraph/resampler/IntegerRatio.h | 54 ++ .../src/flowgraph/resampler/KaiserWindow.h | 87 +++ .../flowgraph/resampler/LinearResampler.cpp | 42 ++ .../src/flowgraph/resampler/LinearResampler.h | 44 ++ .../resampler/MultiChannelResampler.cpp | 171 +++++ .../resampler/MultiChannelResampler.h | 271 ++++++++ .../resampler/PolyphaseResampler.cpp | 61 ++ .../flowgraph/resampler/PolyphaseResampler.h | 51 ++ .../resampler/PolyphaseResamplerMono.cpp | 62 ++ .../resampler/PolyphaseResamplerMono.h | 39 ++ .../resampler/PolyphaseResamplerStereo.cpp | 78 +++ .../resampler/PolyphaseResamplerStereo.h | 39 ++ .../src/flowgraph/resampler/SincResampler.cpp | 76 +++ .../src/flowgraph/resampler/SincResampler.h | 47 ++ .../resampler/SincResamplerStereo.cpp | 80 +++ .../flowgraph/resampler/SincResamplerStereo.h | 39 ++ .../src/opensles/AudioInputStreamOpenSLES.cpp | 352 ++++++++++ .../src/opensles/AudioInputStreamOpenSLES.h | 65 ++ .../opensles/AudioOutputStreamOpenSLES.cpp | 452 +++++++++++++ .../src/opensles/AudioOutputStreamOpenSLES.h | 77 +++ .../oboe/src/opensles/AudioStreamBuffered.cpp | 266 ++++++++ .../oboe/src/opensles/AudioStreamBuffered.h | 92 +++ .../oboe/src/opensles/AudioStreamOpenSLES.cpp | 387 +++++++++++ .../oboe/src/opensles/AudioStreamOpenSLES.h | 132 ++++ .../oboe/src/opensles/EngineOpenSLES.cpp | 101 +++ .../native/oboe/src/opensles/EngineOpenSLES.h | 65 ++ .../oboe/src/opensles/OpenSLESUtilities.cpp | 93 +++ .../oboe/src/opensles/OpenSLESUtilities.h | 44 ++ .../oboe/src/opensles/OutputMixerOpenSLES.cpp | 74 +++ .../oboe/src/opensles/OutputMixerOpenSLES.h | 58 ++ 117 files changed, 13229 insertions(+), 76 deletions(-) create mode 100644 examples/DemoRunner/JuceLibraryCode/AppConfig.h create mode 100644 extras/AudioPluginHost/JuceLibraryCode/AppConfig.h create mode 100644 modules/juce_audio_devices/native/oboe/CMakeLists.txt create mode 100755 modules/juce_audio_devices/native/oboe/LICENSE create mode 100644 modules/juce_audio_devices/native/oboe/include/oboe/AudioStream.h create mode 100644 modules/juce_audio_devices/native/oboe/include/oboe/AudioStreamBase.h create mode 100644 modules/juce_audio_devices/native/oboe/include/oboe/AudioStreamBuilder.h create mode 100644 modules/juce_audio_devices/native/oboe/include/oboe/AudioStreamCallback.h create mode 100644 modules/juce_audio_devices/native/oboe/include/oboe/Definitions.h create mode 100644 modules/juce_audio_devices/native/oboe/include/oboe/LatencyTuner.h create mode 100644 modules/juce_audio_devices/native/oboe/include/oboe/Oboe.h create mode 100644 modules/juce_audio_devices/native/oboe/include/oboe/ResultWithValue.h create mode 100644 modules/juce_audio_devices/native/oboe/include/oboe/StabilizedCallback.h create mode 100644 modules/juce_audio_devices/native/oboe/include/oboe/Utilities.h create mode 100644 modules/juce_audio_devices/native/oboe/include/oboe/Version.h create mode 100644 modules/juce_audio_devices/native/oboe/readme.md create mode 100644 modules/juce_audio_devices/native/oboe/src/aaudio/AAudioLoader.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/aaudio/AAudioLoader.h create mode 100644 modules/juce_audio_devices/native/oboe/src/aaudio/AudioStreamAAudio.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/aaudio/AudioStreamAAudio.h create mode 100644 modules/juce_audio_devices/native/oboe/src/common/AudioClock.h create mode 100644 modules/juce_audio_devices/native/oboe/src/common/AudioSourceCaller.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/common/AudioSourceCaller.h create mode 100644 modules/juce_audio_devices/native/oboe/src/common/AudioStream.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/common/AudioStreamBuilder.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/common/DataConversionFlowGraph.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/common/DataConversionFlowGraph.h create mode 100644 modules/juce_audio_devices/native/oboe/src/common/FilterAudioStream.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/common/FilterAudioStream.h create mode 100644 modules/juce_audio_devices/native/oboe/src/common/FixedBlockAdapter.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/common/FixedBlockAdapter.h create mode 100644 modules/juce_audio_devices/native/oboe/src/common/FixedBlockReader.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/common/FixedBlockReader.h create mode 100644 modules/juce_audio_devices/native/oboe/src/common/FixedBlockWriter.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/common/FixedBlockWriter.h create mode 100644 modules/juce_audio_devices/native/oboe/src/common/LatencyTuner.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/common/MonotonicCounter.h create mode 100644 modules/juce_audio_devices/native/oboe/src/common/OboeDebug.h create mode 100644 modules/juce_audio_devices/native/oboe/src/common/QuirksManager.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/common/QuirksManager.h create mode 100644 modules/juce_audio_devices/native/oboe/src/common/SourceFloatCaller.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/common/SourceFloatCaller.h create mode 100644 modules/juce_audio_devices/native/oboe/src/common/SourceI16Caller.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/common/SourceI16Caller.h create mode 100644 modules/juce_audio_devices/native/oboe/src/common/StabilizedCallback.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/common/Trace.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/common/Trace.h create mode 100644 modules/juce_audio_devices/native/oboe/src/common/Utilities.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/common/Version.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/fifo/FifoBuffer.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/fifo/FifoBuffer.h create mode 100644 modules/juce_audio_devices/native/oboe/src/fifo/FifoController.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/fifo/FifoController.h create mode 100644 modules/juce_audio_devices/native/oboe/src/fifo/FifoControllerBase.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/fifo/FifoControllerBase.h create mode 100644 modules/juce_audio_devices/native/oboe/src/fifo/FifoControllerIndirect.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/fifo/FifoControllerIndirect.h create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/ClipToRange.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/ClipToRange.h create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/FlowGraphNode.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/FlowGraphNode.h create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/ManyToMultiConverter.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/ManyToMultiConverter.h create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/MonoToMultiConverter.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/MonoToMultiConverter.h create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/RampLinear.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/RampLinear.h create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/SampleRateConverter.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/SampleRateConverter.h create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/SinkFloat.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/SinkFloat.h create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/SinkI16.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/SinkI16.h create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/SinkI24.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/SinkI24.h create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/SourceFloat.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/SourceFloat.h create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/SourceI16.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/SourceI16.h create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/SourceI24.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/SourceI24.h create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/HyperbolicCosineWindow.h create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/IntegerRatio.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/IntegerRatio.h create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/KaiserWindow.h create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/LinearResampler.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/LinearResampler.h create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/MultiChannelResampler.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/MultiChannelResampler.h create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/PolyphaseResampler.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/PolyphaseResampler.h create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/PolyphaseResamplerMono.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/PolyphaseResamplerMono.h create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/PolyphaseResamplerStereo.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/PolyphaseResamplerStereo.h create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/SincResampler.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/SincResampler.h create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/SincResamplerStereo.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/SincResamplerStereo.h create mode 100644 modules/juce_audio_devices/native/oboe/src/opensles/AudioInputStreamOpenSLES.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/opensles/AudioInputStreamOpenSLES.h create mode 100644 modules/juce_audio_devices/native/oboe/src/opensles/AudioOutputStreamOpenSLES.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/opensles/AudioOutputStreamOpenSLES.h create mode 100644 modules/juce_audio_devices/native/oboe/src/opensles/AudioStreamBuffered.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/opensles/AudioStreamBuffered.h create mode 100644 modules/juce_audio_devices/native/oboe/src/opensles/AudioStreamOpenSLES.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/opensles/AudioStreamOpenSLES.h create mode 100644 modules/juce_audio_devices/native/oboe/src/opensles/EngineOpenSLES.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/opensles/EngineOpenSLES.h create mode 100644 modules/juce_audio_devices/native/oboe/src/opensles/OpenSLESUtilities.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/opensles/OpenSLESUtilities.h create mode 100644 modules/juce_audio_devices/native/oboe/src/opensles/OutputMixerOpenSLES.cpp create mode 100644 modules/juce_audio_devices/native/oboe/src/opensles/OutputMixerOpenSLES.h diff --git a/examples/DemoRunner/JuceLibraryCode/AppConfig.h b/examples/DemoRunner/JuceLibraryCode/AppConfig.h new file mode 100644 index 0000000000..58349e02ab --- /dev/null +++ b/examples/DemoRunner/JuceLibraryCode/AppConfig.h @@ -0,0 +1,324 @@ +/* + + IMPORTANT! This file is auto-generated each time you save your + project - if you alter its contents, your changes may be overwritten! + + There's a section below where you can add your own custom code safely, and the + Projucer will preserve the contents of that block, but the best way to change + any of these definitions is by using the Projucer's project settings. + + Any commented-out settings will assume their default values. + +*/ + +#pragma once + +//============================================================================== +// [BEGIN_USER_CODE_SECTION] + +// (You can add your own code in this section, and the Projucer will not overwrite it) + +// [END_USER_CODE_SECTION] + +/* + ============================================================================== + + In accordance with the terms of the JUCE 5 End-Use License Agreement, the + JUCE Code in SECTION A cannot be removed, changed or otherwise rendered + ineffective unless you have a JUCE Indie or Pro license, or are using JUCE + under the GPL v3 license. + + End User License Agreement: www.juce.com/juce-5-licence + + ============================================================================== +*/ + +// BEGIN SECTION A + +#ifndef JUCE_DISPLAY_SPLASH_SCREEN + #define JUCE_DISPLAY_SPLASH_SCREEN 0 +#endif + +#ifndef JUCE_REPORT_APP_USAGE + #define JUCE_REPORT_APP_USAGE 0 +#endif + +// END SECTION A + +#define JUCE_USE_DARK_SPLASH_SCREEN 1 + +#define JUCE_PROJUCER_VERSION 0x50407 + +//============================================================================== +#define JUCE_MODULE_AVAILABLE_juce_analytics 1 +#define JUCE_MODULE_AVAILABLE_juce_audio_basics 1 +#define JUCE_MODULE_AVAILABLE_juce_audio_devices 1 +#define JUCE_MODULE_AVAILABLE_juce_audio_formats 1 +#define JUCE_MODULE_AVAILABLE_juce_audio_processors 1 +#define JUCE_MODULE_AVAILABLE_juce_audio_utils 1 +#define JUCE_MODULE_AVAILABLE_juce_blocks_basics 1 +#define JUCE_MODULE_AVAILABLE_juce_box2d 1 +#define JUCE_MODULE_AVAILABLE_juce_core 1 +#define JUCE_MODULE_AVAILABLE_juce_cryptography 1 +#define JUCE_MODULE_AVAILABLE_juce_data_structures 1 +#define JUCE_MODULE_AVAILABLE_juce_dsp 1 +#define JUCE_MODULE_AVAILABLE_juce_events 1 +#define JUCE_MODULE_AVAILABLE_juce_graphics 1 +#define JUCE_MODULE_AVAILABLE_juce_gui_basics 1 +#define JUCE_MODULE_AVAILABLE_juce_gui_extra 1 +#define JUCE_MODULE_AVAILABLE_juce_opengl 1 +#define JUCE_MODULE_AVAILABLE_juce_osc 1 +#define JUCE_MODULE_AVAILABLE_juce_product_unlocking 1 +#define JUCE_MODULE_AVAILABLE_juce_video 1 + +#define JUCE_GLOBAL_MODULE_SETTINGS_INCLUDED 1 + +//============================================================================== +// juce_audio_devices flags: + +#ifndef JUCE_USE_WINRT_MIDI + //#define JUCE_USE_WINRT_MIDI 0 +#endif + +#ifndef JUCE_ASIO + //#define JUCE_ASIO 0 +#endif + +#ifndef JUCE_WASAPI + //#define JUCE_WASAPI 1 +#endif + +#ifndef JUCE_WASAPI_EXCLUSIVE + //#define JUCE_WASAPI_EXCLUSIVE 0 +#endif + +#ifndef JUCE_DIRECTSOUND + //#define JUCE_DIRECTSOUND 1 +#endif + +#ifndef JUCE_ALSA + //#define JUCE_ALSA 1 +#endif + +#ifndef JUCE_JACK + //#define JUCE_JACK 0 +#endif + +#ifndef JUCE_BELA + //#define JUCE_BELA 0 +#endif + +#ifndef JUCE_USE_ANDROID_OBOE + #define JUCE_USE_ANDROID_OBOE 1 +#endif + +#ifndef JUCE_USE_ANDROID_OPENSLES + //#define JUCE_USE_ANDROID_OPENSLES 0 +#endif + +#ifndef JUCE_DISABLE_AUDIO_MIXING_WITH_OTHER_APPS + //#define JUCE_DISABLE_AUDIO_MIXING_WITH_OTHER_APPS 0 +#endif + +//============================================================================== +// juce_audio_formats flags: + +#ifndef JUCE_USE_FLAC + //#define JUCE_USE_FLAC 1 +#endif + +#ifndef JUCE_USE_OGGVORBIS + //#define JUCE_USE_OGGVORBIS 1 +#endif + +#ifndef JUCE_USE_MP3AUDIOFORMAT + #define JUCE_USE_MP3AUDIOFORMAT 1 +#endif + +#ifndef JUCE_USE_LAME_AUDIO_FORMAT + //#define JUCE_USE_LAME_AUDIO_FORMAT 0 +#endif + +#ifndef JUCE_USE_WINDOWS_MEDIA_FORMAT + //#define JUCE_USE_WINDOWS_MEDIA_FORMAT 1 +#endif + +//============================================================================== +// juce_audio_processors flags: + +#ifndef JUCE_PLUGINHOST_VST + //#define JUCE_PLUGINHOST_VST 0 +#endif + +#ifndef JUCE_PLUGINHOST_VST3 + //#define JUCE_PLUGINHOST_VST3 0 +#endif + +#ifndef JUCE_PLUGINHOST_AU + //#define JUCE_PLUGINHOST_AU 0 +#endif + +#ifndef JUCE_PLUGINHOST_LADSPA + //#define JUCE_PLUGINHOST_LADSPA 0 +#endif + +//============================================================================== +// juce_audio_utils flags: + +#ifndef JUCE_USE_CDREADER + //#define JUCE_USE_CDREADER 0 +#endif + +#ifndef JUCE_USE_CDBURNER + //#define JUCE_USE_CDBURNER 0 +#endif + +//============================================================================== +// juce_core flags: + +#ifndef JUCE_FORCE_DEBUG + //#define JUCE_FORCE_DEBUG 0 +#endif + +#ifndef JUCE_LOG_ASSERTIONS + //#define JUCE_LOG_ASSERTIONS 0 +#endif + +#ifndef JUCE_CHECK_MEMORY_LEAKS + //#define JUCE_CHECK_MEMORY_LEAKS 1 +#endif + +#ifndef JUCE_DONT_AUTOLINK_TO_WIN32_LIBRARIES + //#define JUCE_DONT_AUTOLINK_TO_WIN32_LIBRARIES 0 +#endif + +#ifndef JUCE_INCLUDE_ZLIB_CODE + //#define JUCE_INCLUDE_ZLIB_CODE 1 +#endif + +#ifndef JUCE_USE_CURL + //#define JUCE_USE_CURL 1 +#endif + +#ifndef JUCE_LOAD_CURL_SYMBOLS_LAZILY + //#define JUCE_LOAD_CURL_SYMBOLS_LAZILY 0 +#endif + +#ifndef JUCE_CATCH_UNHANDLED_EXCEPTIONS + //#define JUCE_CATCH_UNHANDLED_EXCEPTIONS 0 +#endif + +#ifndef JUCE_ALLOW_STATIC_NULL_VARIABLES + #define JUCE_ALLOW_STATIC_NULL_VARIABLES 0 +#endif + +#ifndef JUCE_STRICT_REFCOUNTEDPOINTER + #define JUCE_STRICT_REFCOUNTEDPOINTER 1 +#endif + +//============================================================================== +// juce_dsp flags: + +#ifndef JUCE_ASSERTION_FIRFILTER + //#define JUCE_ASSERTION_FIRFILTER 1 +#endif + +#ifndef JUCE_DSP_USE_INTEL_MKL + //#define JUCE_DSP_USE_INTEL_MKL 0 +#endif + +#ifndef JUCE_DSP_USE_SHARED_FFTW + //#define JUCE_DSP_USE_SHARED_FFTW 0 +#endif + +#ifndef JUCE_DSP_USE_STATIC_FFTW + //#define JUCE_DSP_USE_STATIC_FFTW 0 +#endif + +#ifndef JUCE_DSP_ENABLE_SNAP_TO_ZERO + //#define JUCE_DSP_ENABLE_SNAP_TO_ZERO 1 +#endif + +//============================================================================== +// juce_events flags: + +#ifndef JUCE_EXECUTE_APP_SUSPEND_ON_BACKGROUND_TASK + //#define JUCE_EXECUTE_APP_SUSPEND_ON_BACKGROUND_TASK 0 +#endif + +//============================================================================== +// juce_graphics flags: + +#ifndef JUCE_USE_COREIMAGE_LOADER + //#define JUCE_USE_COREIMAGE_LOADER 1 +#endif + +#ifndef JUCE_USE_DIRECTWRITE + //#define JUCE_USE_DIRECTWRITE 1 +#endif + +#ifndef JUCE_DISABLE_COREGRAPHICS_FONT_SMOOTHING + //#define JUCE_DISABLE_COREGRAPHICS_FONT_SMOOTHING 0 +#endif + +//============================================================================== +// juce_gui_basics flags: + +#ifndef JUCE_ENABLE_REPAINT_DEBUGGING + //#define JUCE_ENABLE_REPAINT_DEBUGGING 0 +#endif + +#ifndef JUCE_USE_XRANDR + //#define JUCE_USE_XRANDR 1 +#endif + +#ifndef JUCE_USE_XINERAMA + //#define JUCE_USE_XINERAMA 1 +#endif + +#ifndef JUCE_USE_XSHM + //#define JUCE_USE_XSHM 1 +#endif + +#ifndef JUCE_USE_XRENDER + //#define JUCE_USE_XRENDER 0 +#endif + +#ifndef JUCE_USE_XCURSOR + //#define JUCE_USE_XCURSOR 1 +#endif + +#ifndef JUCE_WIN_PER_MONITOR_DPI_AWARE + //#define JUCE_WIN_PER_MONITOR_DPI_AWARE 1 +#endif + +//============================================================================== +// juce_gui_extra flags: + +#ifndef JUCE_WEB_BROWSER + //#define JUCE_WEB_BROWSER 1 +#endif + +#ifndef JUCE_ENABLE_LIVE_CONSTANT_EDITOR + //#define JUCE_ENABLE_LIVE_CONSTANT_EDITOR 0 +#endif + +//============================================================================== +// juce_video flags: + +#ifndef JUCE_USE_CAMERA + #define JUCE_USE_CAMERA 1 +#endif + +#ifndef JUCE_SYNC_VIDEO_VOLUME_WITH_OS_MEDIA_VOLUME + //#define JUCE_SYNC_VIDEO_VOLUME_WITH_OS_MEDIA_VOLUME 1 +#endif + +//============================================================================== +#ifndef JUCE_STANDALONE_APPLICATION + #if defined(JucePlugin_Name) && defined(JucePlugin_Build_Standalone) + #define JUCE_STANDALONE_APPLICATION JucePlugin_Build_Standalone + #else + #define JUCE_STANDALONE_APPLICATION 1 + #endif +#endif diff --git a/extras/AudioPluginHost/JuceLibraryCode/AppConfig.h b/extras/AudioPluginHost/JuceLibraryCode/AppConfig.h new file mode 100644 index 0000000000..6a6ca936ee --- /dev/null +++ b/extras/AudioPluginHost/JuceLibraryCode/AppConfig.h @@ -0,0 +1,297 @@ +/* + + IMPORTANT! This file is auto-generated each time you save your + project - if you alter its contents, your changes may be overwritten! + + There's a section below where you can add your own custom code safely, and the + Projucer will preserve the contents of that block, but the best way to change + any of these definitions is by using the Projucer's project settings. + + Any commented-out settings will assume their default values. + +*/ + +#pragma once + +//============================================================================== +// [BEGIN_USER_CODE_SECTION] + +#ifndef JUCE_ANDROID + #define JUCE_MODAL_LOOPS_PERMITTED (! JUCE_IOS) +#endif + +// [END_USER_CODE_SECTION] + +/* + ============================================================================== + + In accordance with the terms of the JUCE 5 End-Use License Agreement, the + JUCE Code in SECTION A cannot be removed, changed or otherwise rendered + ineffective unless you have a JUCE Indie or Pro license, or are using JUCE + under the GPL v3 license. + + End User License Agreement: www.juce.com/juce-5-licence + + ============================================================================== +*/ + +// BEGIN SECTION A + +#ifndef JUCE_DISPLAY_SPLASH_SCREEN + #define JUCE_DISPLAY_SPLASH_SCREEN 0 +#endif + +#ifndef JUCE_REPORT_APP_USAGE + #define JUCE_REPORT_APP_USAGE 0 +#endif + +// END SECTION A + +#define JUCE_USE_DARK_SPLASH_SCREEN 1 + +#define JUCE_PROJUCER_VERSION 0x50407 + +//============================================================================== +#define JUCE_MODULE_AVAILABLE_juce_audio_basics 1 +#define JUCE_MODULE_AVAILABLE_juce_audio_devices 1 +#define JUCE_MODULE_AVAILABLE_juce_audio_formats 1 +#define JUCE_MODULE_AVAILABLE_juce_audio_processors 1 +#define JUCE_MODULE_AVAILABLE_juce_audio_utils 1 +#define JUCE_MODULE_AVAILABLE_juce_core 1 +#define JUCE_MODULE_AVAILABLE_juce_cryptography 1 +#define JUCE_MODULE_AVAILABLE_juce_data_structures 1 +#define JUCE_MODULE_AVAILABLE_juce_events 1 +#define JUCE_MODULE_AVAILABLE_juce_graphics 1 +#define JUCE_MODULE_AVAILABLE_juce_gui_basics 1 +#define JUCE_MODULE_AVAILABLE_juce_gui_extra 1 +#define JUCE_MODULE_AVAILABLE_juce_opengl 1 +#define JUCE_MODULE_AVAILABLE_juce_video 1 + +#define JUCE_GLOBAL_MODULE_SETTINGS_INCLUDED 1 + +//============================================================================== +// juce_audio_devices flags: + +#ifndef JUCE_USE_WINRT_MIDI + //#define JUCE_USE_WINRT_MIDI 0 +#endif + +#ifndef JUCE_ASIO + //#define JUCE_ASIO 0 +#endif + +#ifndef JUCE_WASAPI + #define JUCE_WASAPI 1 +#endif + +#ifndef JUCE_WASAPI_EXCLUSIVE + //#define JUCE_WASAPI_EXCLUSIVE 0 +#endif + +#ifndef JUCE_DIRECTSOUND + #define JUCE_DIRECTSOUND 1 +#endif + +#ifndef JUCE_ALSA + #define JUCE_ALSA 1 +#endif + +#ifndef JUCE_JACK + //#define JUCE_JACK 0 +#endif + +#ifndef JUCE_BELA + //#define JUCE_BELA 0 +#endif + +#ifndef JUCE_USE_ANDROID_OBOE + #define JUCE_USE_ANDROID_OBOE 1 +#endif + +#ifndef JUCE_USE_ANDROID_OPENSLES + //#define JUCE_USE_ANDROID_OPENSLES 0 +#endif + +#ifndef JUCE_DISABLE_AUDIO_MIXING_WITH_OTHER_APPS + //#define JUCE_DISABLE_AUDIO_MIXING_WITH_OTHER_APPS 0 +#endif + +//============================================================================== +// juce_audio_formats flags: + +#ifndef JUCE_USE_FLAC + #define JUCE_USE_FLAC 0 +#endif + +#ifndef JUCE_USE_OGGVORBIS + #define JUCE_USE_OGGVORBIS 0 +#endif + +#ifndef JUCE_USE_MP3AUDIOFORMAT + //#define JUCE_USE_MP3AUDIOFORMAT 0 +#endif + +#ifndef JUCE_USE_LAME_AUDIO_FORMAT + //#define JUCE_USE_LAME_AUDIO_FORMAT 0 +#endif + +#ifndef JUCE_USE_WINDOWS_MEDIA_FORMAT + //#define JUCE_USE_WINDOWS_MEDIA_FORMAT 1 +#endif + +//============================================================================== +// juce_audio_processors flags: + +#ifndef JUCE_PLUGINHOST_VST + //#define JUCE_PLUGINHOST_VST 0 +#endif + +#ifndef JUCE_PLUGINHOST_VST3 + #define JUCE_PLUGINHOST_VST3 1 +#endif + +#ifndef JUCE_PLUGINHOST_AU + #define JUCE_PLUGINHOST_AU 1 +#endif + +#ifndef JUCE_PLUGINHOST_LADSPA + #define JUCE_PLUGINHOST_LADSPA 1 +#endif + +//============================================================================== +// juce_audio_utils flags: + +#ifndef JUCE_USE_CDREADER + #define JUCE_USE_CDREADER 0 +#endif + +#ifndef JUCE_USE_CDBURNER + #define JUCE_USE_CDBURNER 0 +#endif + +//============================================================================== +// juce_core flags: + +#ifndef JUCE_FORCE_DEBUG + //#define JUCE_FORCE_DEBUG 0 +#endif + +#ifndef JUCE_LOG_ASSERTIONS + //#define JUCE_LOG_ASSERTIONS 0 +#endif + +#ifndef JUCE_CHECK_MEMORY_LEAKS + //#define JUCE_CHECK_MEMORY_LEAKS 1 +#endif + +#ifndef JUCE_DONT_AUTOLINK_TO_WIN32_LIBRARIES + //#define JUCE_DONT_AUTOLINK_TO_WIN32_LIBRARIES 0 +#endif + +#ifndef JUCE_INCLUDE_ZLIB_CODE + //#define JUCE_INCLUDE_ZLIB_CODE 1 +#endif + +#ifndef JUCE_USE_CURL + //#define JUCE_USE_CURL 1 +#endif + +#ifndef JUCE_LOAD_CURL_SYMBOLS_LAZILY + //#define JUCE_LOAD_CURL_SYMBOLS_LAZILY 0 +#endif + +#ifndef JUCE_CATCH_UNHANDLED_EXCEPTIONS + //#define JUCE_CATCH_UNHANDLED_EXCEPTIONS 0 +#endif + +#ifndef JUCE_ALLOW_STATIC_NULL_VARIABLES + //#define JUCE_ALLOW_STATIC_NULL_VARIABLES 0 +#endif + +#ifndef JUCE_STRICT_REFCOUNTEDPOINTER + //#define JUCE_STRICT_REFCOUNTEDPOINTER 0 +#endif + +//============================================================================== +// juce_events flags: + +#ifndef JUCE_EXECUTE_APP_SUSPEND_ON_BACKGROUND_TASK + //#define JUCE_EXECUTE_APP_SUSPEND_ON_BACKGROUND_TASK 0 +#endif + +//============================================================================== +// juce_graphics flags: + +#ifndef JUCE_USE_COREIMAGE_LOADER + //#define JUCE_USE_COREIMAGE_LOADER 1 +#endif + +#ifndef JUCE_USE_DIRECTWRITE + //#define JUCE_USE_DIRECTWRITE 1 +#endif + +#ifndef JUCE_DISABLE_COREGRAPHICS_FONT_SMOOTHING + //#define JUCE_DISABLE_COREGRAPHICS_FONT_SMOOTHING 0 +#endif + +//============================================================================== +// juce_gui_basics flags: + +#ifndef JUCE_ENABLE_REPAINT_DEBUGGING + //#define JUCE_ENABLE_REPAINT_DEBUGGING 0 +#endif + +#ifndef JUCE_USE_XRANDR + //#define JUCE_USE_XRANDR 1 +#endif + +#ifndef JUCE_USE_XINERAMA + //#define JUCE_USE_XINERAMA 1 +#endif + +#ifndef JUCE_USE_XSHM + //#define JUCE_USE_XSHM 1 +#endif + +#ifndef JUCE_USE_XRENDER + //#define JUCE_USE_XRENDER 0 +#endif + +#ifndef JUCE_USE_XCURSOR + //#define JUCE_USE_XCURSOR 1 +#endif + +#ifndef JUCE_WIN_PER_MONITOR_DPI_AWARE + //#define JUCE_WIN_PER_MONITOR_DPI_AWARE 1 +#endif + +//============================================================================== +// juce_gui_extra flags: + +#ifndef JUCE_WEB_BROWSER + #define JUCE_WEB_BROWSER 0 +#endif + +#ifndef JUCE_ENABLE_LIVE_CONSTANT_EDITOR + //#define JUCE_ENABLE_LIVE_CONSTANT_EDITOR 0 +#endif + +//============================================================================== +// juce_video flags: + +#ifndef JUCE_USE_CAMERA + #define JUCE_USE_CAMERA 0 +#endif + +#ifndef JUCE_SYNC_VIDEO_VOLUME_WITH_OS_MEDIA_VOLUME + //#define JUCE_SYNC_VIDEO_VOLUME_WITH_OS_MEDIA_VOLUME 1 +#endif + +//============================================================================== +#ifndef JUCE_STANDALONE_APPLICATION + #if defined(JucePlugin_Name) && defined(JucePlugin_Build_Standalone) + #define JUCE_STANDALONE_APPLICATION JucePlugin_Build_Standalone + #else + #define JUCE_STANDALONE_APPLICATION 1 + #endif +#endif diff --git a/extras/Projucer/Source/ProjectSaving/jucer_ProjectExport_Android.h b/extras/Projucer/Source/ProjectSaving/jucer_ProjectExport_Android.h index 796c55b350..b00f37fa74 100644 --- a/extras/Projucer/Source/ProjectSaving/jucer_ProjectExport_Android.h +++ b/extras/Projucer/Source/ProjectSaving/jucer_ProjectExport_Android.h @@ -341,15 +341,32 @@ private: mo << "cmake_minimum_required(VERSION 3.4.1)" << newLine << newLine; if (! isLibrary()) - mo << "SET(BINARY_NAME \"juce_jni\")" << newLine << newLine; + mo << "set(BINARY_NAME \"juce_jni\")" << newLine << newLine; - auto useOboe = project.getEnabledModules().isModuleEnabled ("juce_audio_devices") && project.isConfigFlagEnabled ("JUCE_USE_ANDROID_OBOE", false); + auto useOboe = project.getEnabledModules().isModuleEnabled ("juce_audio_devices") + && project.isConfigFlagEnabled ("JUCE_USE_ANDROID_OBOE", true); if (useOboe) { - String oboePath (androidOboeRepositoryPath.get().toString().trim().quoted()); + auto oboePath = [&] + { + auto oboeDir = androidOboeRepositoryPath.get().toString().trim(); - mo << "SET(OBOE_DIR " << oboePath << ")" << newLine << newLine; + if (oboeDir.isEmpty()) + oboeDir = getModuleFolderRelativeToProject ("juce_audio_devices").getChildFile ("native") + .getChildFile ("oboe") + .rebased (getProject().getProjectFolder(), getTargetFolder(), + build_tools::RelativePath::buildTargetFolder) + .toUnixStyle(); + + // CMakeLists.txt is in the "app" subfolder + if (! build_tools::isAbsolutePath (oboeDir)) + oboeDir = "../" + oboeDir; + + return expandHomeFolderToken (oboeDir); + }(); + + mo << "set(OBOE_DIR " << oboePath.quoted() << ")" << newLine << newLine; mo << "add_subdirectory (${OBOE_DIR} ./oboe)" << newLine << newLine; } @@ -371,17 +388,14 @@ private: mo << " \"${ANDROID_NDK}/sources/android/cpufeatures\"" << newLine; - if (useOboe) - mo << " \"${OBOE_DIR}/include\"" << newLine; - mo << ")" << newLine << newLine; } auto cfgExtraLinkerFlags = getExtraLinkerFlagsString(); if (cfgExtraLinkerFlags.isNotEmpty()) { - mo << "SET( JUCE_LDFLAGS \"" << cfgExtraLinkerFlags.replace ("\"", "\\\"") << "\")" << newLine; - mo << "SET( CMAKE_SHARED_LINKER_FLAGS \"${CMAKE_EXE_LINKER_FLAGS} ${JUCE_LDFLAGS}\")" << newLine << newLine; + mo << "set( JUCE_LDFLAGS \"" << cfgExtraLinkerFlags.replace ("\"", "\\\"") << "\")" << newLine; + mo << "set( CMAKE_SHARED_LINKER_FLAGS \"${CMAKE_EXE_LINKER_FLAGS} ${JUCE_LDFLAGS}\")" << newLine << newLine; } mo << "enable_language(ASM)" << newLine << newLine; @@ -405,11 +419,11 @@ private: && cfgHeaderPaths.size() == 0 && cfgLibraryPaths.size() == 0) continue; - mo << (first ? "IF" : "ELSEIF") << "(JUCE_BUILD_CONFIGURATION MATCHES \"" << cfg.getProductFlavourCMakeIdentifier() <<"\")" << newLine; + mo << (first ? "if" : "elseif") << "(JUCE_BUILD_CONFIGURATION MATCHES \"" << cfg.getProductFlavourCMakeIdentifier() <<"\")" << newLine; if (isLibrary()) { - mo << " SET(BINARY_NAME \"" << getNativeModuleBinaryName (cfg) << "\")" << newLine; + mo << " set(BINARY_NAME \"" << getNativeModuleBinaryName (cfg) << "\")" << newLine; auto binaryLocation = cfg.getTargetBinaryRelativePathString(); @@ -419,7 +433,7 @@ private: .rebased (getProject().getFile().getParentDirectory(), file.getParentDirectory(), build_tools::RelativePath::buildTargetFolder); - mo << " SET(CMAKE_ARCHIVE_OUTPUT_DIRECTORY \"" << "../../../../" << locationRelativeToCmake.toUnixStyle() << "\")" << newLine; + mo << " set(CMAKE_ARCHIVE_OUTPUT_DIRECTORY \"" << "../../../../" << locationRelativeToCmake.toUnixStyle() << "\")" << newLine; } } @@ -454,10 +468,10 @@ private: for (auto& variable : cmakeVariables) { auto configVariable = variable + "_" + cfg.getProductFlavourCMakeIdentifier(); - mo << " SET(" << configVariable << " \"${" << configVariable << "} -flto\")" << newLine; + mo << " set(" << configVariable << " \"${" << configVariable << "} -flto\")" << newLine; } - mo << " ENDIF(" << mipsCondition << ")" << newLine; + mo << " endif()" << newLine; } first = false; @@ -469,11 +483,11 @@ private: if (config) { - if (auto* cfg = dynamic_cast (config.get())) + if (dynamic_cast (config.get()) != nullptr) { - mo << "ELSE(JUCE_BUILD_CONFIGURATION MATCHES \"" << cfg->getProductFlavourCMakeIdentifier() <<"\")" << newLine; - mo << " MESSAGE( FATAL_ERROR \"No matching build-configuration found.\" )" << newLine; - mo << "ENDIF(JUCE_BUILD_CONFIGURATION MATCHES \"" << cfg->getProductFlavourCMakeIdentifier() <<"\")" << newLine << newLine; + mo << "else()" << newLine; + mo << " message( FATAL_ERROR \"No matching build-configuration found.\" )" << newLine; + mo << "endif()" << newLine << newLine; } } } @@ -505,6 +519,25 @@ private: mo << newLine; } + auto flags = getProjectCompilerFlags(); + + if (flags.size() > 0) + mo << "target_compile_options( ${BINARY_NAME} PRIVATE " << flags.joinIntoString (" ") << " )" << newLine << newLine; + + for (ConstConfigIterator config (*this); config.next();) + { + auto& cfg = dynamic_cast (*config); + + mo << "if( JUCE_BUILD_CONFIGURATION MATCHES \"" << cfg.getProductFlavourCMakeIdentifier() << "\" )" << newLine; + mo << " target_compile_options( ${BINARY_NAME} PRIVATE"; + + for (auto& flag : cfg.getRecommendedCompilerWarningFlags()) + mo << " " << flag; + + mo << ")" << newLine; + mo << "endif()" << newLine << newLine; + } + auto libraries = getAndroidLibraries(); if (libraries.size() > 0) { @@ -642,9 +675,6 @@ private: mo << ", \"-DCMAKE_CXX_FLAGS_" << (cfg.isDebug() ? "DEBUG" : "RELEASE") << "=-O" << cfg.getGCCOptimisationFlag(); - for (auto& flag : cfg.getRecommendedCompilerWarningFlags()) - mo << " " << flag; - mo << "\"" << ", \"-DCMAKE_C_FLAGS_" << (cfg.isDebug() ? "DEBUG" : "RELEASE") << "=-O" << cfg.getGCCOptimisationFlag() @@ -686,8 +716,6 @@ private: { auto bundleIdentifier = project.getBundleIdentifierString().toLowerCase(); auto cmakeDefs = getCmakeDefinitions(); - auto cFlags = getProjectCompilerFlags(); - auto cxxFlags = getProjectCxxCompilerFlags(); auto minSdkVersion = static_cast (androidMinimumSDK.get()); auto targetSdkVersion = static_cast (androidTargetSDK.get()); @@ -707,12 +735,6 @@ private: mo << " arguments " << cmakeDefs.joinIntoString (", ") << newLine; - if (cFlags.size() > 0) - mo << " cFlags " << cFlags.joinIntoString (", ") << newLine; - - if (cxxFlags.size() > 0) - mo << " cppFlags " << cxxFlags.joinIntoString (", ") << newLine; - mo << " }" << newLine; mo << " }" << newLine; mo << " }" << newLine; @@ -1049,9 +1071,10 @@ private: //============================================================================== void createManifestExporterProperties (PropertyListBuilder& props) { - props.add (new TextPropertyComponent (androidOboeRepositoryPath, "Oboe Repository Path", 2048, false), - "Path to the root of Oboe repository. Make sure to point Oboe repository to " - "commit with SHA c5c3cc17f78974bf005bf33a2de1a093ac55cc07 before building."); + props.add (new TextPropertyComponent (androidOboeRepositoryPath, "Custom Oboe Repository", 2048, false), + "Path to the root of Oboe repository. This path can be absolute, or relative to the build directory. " + "Make sure to point Oboe repository to commit with SHA c5c3cc17f78974bf005bf33a2de1a093ac55cc07 before building. " + "Leave blank to use the version of Oboe distributed with JUCE."); props.add (new ChoicePropertyComponent (androidInternetNeeded, "Internet Access"), "If enabled, this will set the android.permission.INTERNET flag in the manifest."); @@ -1374,6 +1397,19 @@ private: cmakeArgs.add ("\"-DANDROID_ARM_MODE=arm\""); cmakeArgs.add ("\"-DANDROID_ARM_NEON=TRUE\""); + auto cppStandard = [this] + { + auto projectStandard = project.getCppStandardString(); + + if (projectStandard == "latest") + return String ("17"); + + return projectStandard; + }(); + + cmakeArgs.add ("\"-DCMAKE_CXX_STANDARD=" + cppStandard + "\""); + cmakeArgs.add ("\"-DCMAKE_CXX_EXTENSIONS=" + String (shouldUseGNUExtensions() ? "ON" : "OFF") + "\""); + return cmakeArgs; } @@ -1386,22 +1422,6 @@ private: return cFlags; } - StringArray getAndroidCxxCompilerFlags() const - { - auto cxxFlags = getAndroidCompilerFlags(); - - auto cppStandard = project.getCppStandardString(); - - if (cppStandard == "latest" || cppStandard == "17") // C++17 flag isn't supported yet so use 1z for now - cppStandard = "1z"; - - cppStandard = "-std=" + String (shouldUseGNUExtensions() ? "gnu++" : "c++") + cppStandard; - - cxxFlags.add (cppStandard.quoted()); - - return cxxFlags; - } - StringArray getProjectCompilerFlags() const { auto cFlags = getAndroidCompilerFlags(); @@ -1409,13 +1429,6 @@ private: return cFlags; } - StringArray getProjectCxxCompilerFlags() const - { - auto cxxFlags = getAndroidCxxCompilerFlags(); - cxxFlags.addArray (getEscapedFlags (StringArray::fromTokens (getExtraCompilerFlagsString(), true))); - return cxxFlags; - } - //============================================================================== StringPairArray getAndroidPreprocessorDefs() const { diff --git a/modules/juce_audio_devices/juce_audio_devices.cpp b/modules/juce_audio_devices/juce_audio_devices.cpp index e0066c2f30..924e01655a 100644 --- a/modules/juce_audio_devices/juce_audio_devices.cpp +++ b/modules/juce_audio_devices/juce_audio_devices.cpp @@ -167,7 +167,11 @@ #error "Oboe cannot be enabled at the same time as openSL! Please disable JUCE_USE_ANDROID_OPENSLES" #endif - JUCE_BEGIN_IGNORE_WARNINGS_GCC_LIKE ("-Wunused-parameter") + JUCE_BEGIN_IGNORE_WARNINGS_GCC_LIKE ("-Wunused-parameter", + "-Wzero-as-null-pointer-constant", + "-Winconsistent-missing-destructor-override", + "-Wshadow-field-in-constructor", + "-Wshadow-field") #include JUCE_END_IGNORE_WARNINGS_GCC_LIKE #endif diff --git a/modules/juce_audio_devices/juce_audio_devices.h b/modules/juce_audio_devices/juce_audio_devices.h index a11b1e7eba..cb641ec512 100644 --- a/modules/juce_audio_devices/juce_audio_devices.h +++ b/modules/juce_audio_devices/juce_audio_devices.h @@ -132,18 +132,10 @@ #endif /** Config: JUCE_USE_ANDROID_OBOE - *** - DEVELOPER PREVIEW - Oboe is currently in developer preview and - is in active development. This preview allows for early access - and evaluation for developers targeting Android platform. - *** - - Enables Oboe devices (Android only, API 16 or above). Requires - Oboe repository path to be specified in Android exporter. + Enables Oboe devices (Android only, API 16 or above). */ - #ifndef JUCE_USE_ANDROID_OBOE - #define JUCE_USE_ANDROID_OBOE 0 + #define JUCE_USE_ANDROID_OBOE 1 #endif #if JUCE_USE_ANDROID_OBOE && JUCE_ANDROID_API_VERSION < 16 diff --git a/modules/juce_audio_devices/native/juce_android_Oboe.cpp b/modules/juce_audio_devices/native/juce_android_Oboe.cpp index eeb5758425..605ffca2b7 100644 --- a/modules/juce_audio_devices/native/juce_android_Oboe.cpp +++ b/modules/juce_audio_devices/native/juce_android_Oboe.cpp @@ -152,7 +152,7 @@ public: jassert (inputDeviceId != -1 || outputDeviceId != -1); } - ~OboeAudioIODevice() + ~OboeAudioIODevice() override { close(); } @@ -420,11 +420,11 @@ private: OboeStream (int deviceId, oboe::Direction direction, oboe::SharingMode sharingMode, int channelCount, oboe::AudioFormat format, - int32 sampleRate, int32 bufferSize, - oboe::AudioStreamCallback* callback = nullptr) + int32 sampleRateIn, int32 bufferSize, + oboe::AudioStreamCallback* callbackIn = nullptr) { open (deviceId, direction, sharingMode, channelCount, - format, sampleRate, bufferSize, callback); + format, sampleRateIn, bufferSize, callbackIn); } ~OboeStream() @@ -692,11 +692,11 @@ private: { public: OboeSessionImpl (OboeAudioIODevice& ownerToUse, - int inputDeviceId, int outputDeviceId, + int inputDeviceIdIn, int outputDeviceIdIn, int numInputChannelsToUse, int numOutputChannelsToUse, int sampleRateToUse, int bufferSizeToUse) : OboeSessionBase (ownerToUse, - inputDeviceId, outputDeviceId, + inputDeviceIdIn, outputDeviceIdIn, numInputChannelsToUse, numOutputChannelsToUse, sampleRateToUse, bufferSizeToUse, OboeAudioIODeviceBufferHelpers::oboeAudioFormat(), @@ -740,7 +740,7 @@ private: if (stream == nullptr || ! openedOk()) return false; - auto result = stream->getNativeStream()->getTimestamp (CLOCK_MONOTONIC, 0, 0); + auto result = stream->getNativeStream()->getTimestamp (CLOCK_MONOTONIC, nullptr, nullptr); return result != oboe::Result::ErrorUnimplemented; } @@ -1135,9 +1135,9 @@ public: jclass audioManagerClass = env->FindClass ("android/media/AudioManager"); // We should be really entering here only if API supports it. - jassert (audioManagerClass != 0); + jassert (audioManagerClass != nullptr); - if (audioManagerClass == 0) + if (audioManagerClass == nullptr) return; auto audioManager = LocalRef (env->CallObjectMethod (getAppContext().get(), @@ -1256,7 +1256,7 @@ public: { auto* env = getEnv(); - jint* jArrayElems = env->GetIntArrayElements (jArray, 0); + jint* jArrayElems = env->GetIntArrayElements (jArray, nullptr); int numElems = env->GetArrayLength (jArray); Array juceArray; diff --git a/modules/juce_audio_devices/native/oboe/CMakeLists.txt b/modules/juce_audio_devices/native/oboe/CMakeLists.txt new file mode 100644 index 0000000000..f05dd460ab --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/CMakeLists.txt @@ -0,0 +1,91 @@ +cmake_minimum_required(VERSION 3.4.1) + +# Set the name of the project and store it in PROJECT_NAME. Also set the following variables: +# PROJECT_SOURCE_DIR (usually the root directory where Oboe has been cloned e.g.) +# PROJECT_BINARY_DIR (usually the containing project's binary directory, +# e.g. ${OBOE_HOME}/samples/RhythmGame/.externalNativeBuild/cmake/ndkExtractorDebug/x86/oboe-bin) +project(oboe) + +set (oboe_sources + src/aaudio/AAudioLoader.cpp + src/aaudio/AudioStreamAAudio.cpp + src/common/AudioSourceCaller.cpp + src/common/AudioStream.cpp + src/common/AudioStreamBuilder.cpp + src/common/DataConversionFlowGraph.cpp + src/common/FilterAudioStream.cpp + src/common/FixedBlockAdapter.cpp + src/common/FixedBlockReader.cpp + src/common/FixedBlockWriter.cpp + src/common/LatencyTuner.cpp + src/common/SourceFloatCaller.cpp + src/common/SourceI16Caller.cpp + src/common/Utilities.cpp + src/common/QuirksManager.cpp + src/fifo/FifoBuffer.cpp + src/fifo/FifoController.cpp + src/fifo/FifoControllerBase.cpp + src/fifo/FifoControllerIndirect.cpp + src/flowgraph/FlowGraphNode.cpp + src/flowgraph/ClipToRange.cpp + src/flowgraph/ManyToMultiConverter.cpp + src/flowgraph/MonoToMultiConverter.cpp + src/flowgraph/RampLinear.cpp + src/flowgraph/SampleRateConverter.cpp + src/flowgraph/SinkFloat.cpp + src/flowgraph/SinkI16.cpp + src/flowgraph/SinkI24.cpp + src/flowgraph/SourceFloat.cpp + src/flowgraph/SourceI16.cpp + src/flowgraph/SourceI24.cpp + src/flowgraph/resampler/IntegerRatio.cpp + src/flowgraph/resampler/LinearResampler.cpp + src/flowgraph/resampler/MultiChannelResampler.cpp + src/flowgraph/resampler/PolyphaseResampler.cpp + src/flowgraph/resampler/PolyphaseResamplerMono.cpp + src/flowgraph/resampler/PolyphaseResamplerStereo.cpp + src/flowgraph/resampler/SincResampler.cpp + src/flowgraph/resampler/SincResamplerStereo.cpp + src/opensles/AudioInputStreamOpenSLES.cpp + src/opensles/AudioOutputStreamOpenSLES.cpp + src/opensles/AudioStreamBuffered.cpp + src/opensles/AudioStreamOpenSLES.cpp + src/opensles/EngineOpenSLES.cpp + src/opensles/OpenSLESUtilities.cpp + src/opensles/OutputMixerOpenSLES.cpp + src/common/StabilizedCallback.cpp + src/common/Trace.cpp + src/common/Version.cpp + ) + +add_library(oboe ${oboe_sources}) + +# Specify directories which the compiler should look for headers +target_include_directories(oboe + PRIVATE src + PUBLIC include) + +# JUCE CHANGE STARTS HERE + +# This comment provided for Apache License compliance. We've removed the extra warnings flags and +# the `-Werror` option, to avoid cases where compilers produce unexpected errors and fail the build. +# We've also removed the explicit `-std=c++14` compile option, and replaced it with a more +# cmake-friendly way of specifying the language standard. + +target_compile_options(oboe PRIVATE -Ofast) +set_target_properties(oboe PROPERTIES CXX_STANDARD 14 CXX_STANDARD_REQUIRED TRUE CXX_EXTENSIONS FALSE) + +# JUCE CHANGE ENDS HERE + +# Enable logging for debug builds +target_compile_definitions(oboe PUBLIC $<$:OBOE_ENABLE_LOGGING=1>) + +target_link_libraries(oboe PRIVATE log OpenSLES) + +# When installing oboe put the libraries in the lib/ folder e.g. lib/arm64-v8a +install(TARGETS oboe + LIBRARY DESTINATION lib/${ANDROID_ABI} + ARCHIVE DESTINATION lib/${ANDROID_ABI}) + +# Also install the headers +install(DIRECTORY include/oboe DESTINATION include) diff --git a/modules/juce_audio_devices/native/oboe/LICENSE b/modules/juce_audio_devices/native/oboe/LICENSE new file mode 100755 index 0000000000..d645695673 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/LICENSE @@ -0,0 +1,202 @@ + + Apache License + Version 2.0, January 2004 + http://www.apache.org/licenses/ + + TERMS AND CONDITIONS FOR USE, REPRODUCTION, AND DISTRIBUTION + + 1. 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Might do more in the future. + } + + /** + * Close the stream and deallocate any resources from the open() call. + */ + virtual Result close(); + + /** + * Start the stream. This will block until the stream has been started, an error occurs + * or `timeoutNanoseconds` has been reached. + */ + virtual Result start(int64_t timeoutNanoseconds = kDefaultTimeoutNanos); + + /** + * Pause the stream. This will block until the stream has been paused, an error occurs + * or `timeoutNanoseconds` has been reached. + */ + virtual Result pause(int64_t timeoutNanoseconds = kDefaultTimeoutNanos); + + /** + * Flush the stream. This will block until the stream has been flushed, an error occurs + * or `timeoutNanoseconds` has been reached. + */ + virtual Result flush(int64_t timeoutNanoseconds = kDefaultTimeoutNanos); + + /** + * Stop the stream. This will block until the stream has been stopped, an error occurs + * or `timeoutNanoseconds` has been reached. + */ + virtual Result stop(int64_t timeoutNanoseconds = kDefaultTimeoutNanos); + + /* Asynchronous requests. + * Use waitForStateChange() if you need to wait for completion. + */ + + /** + * Start the stream asynchronously. Returns immediately (does not block). Equivalent to calling + * `start(0)`. + */ + virtual Result requestStart() = 0; + + /** + * Pause the stream asynchronously. Returns immediately (does not block). Equivalent to calling + * `pause(0)`. + */ + virtual Result requestPause() = 0; + + /** + * Flush the stream asynchronously. Returns immediately (does not block). Equivalent to calling + * `flush(0)`. + */ + virtual Result requestFlush() = 0; + + /** + * Stop the stream asynchronously. Returns immediately (does not block). Equivalent to calling + * `stop(0)`. + */ + virtual Result requestStop() = 0; + + /** + * Query the current state, eg. StreamState::Pausing + * + * @return state or a negative error. + */ + virtual StreamState getState() const = 0; + + /** + * Wait until the stream's current state no longer matches the input state. + * The input state is passed to avoid race conditions caused by the state + * changing between calls. + * + * Note that generally applications do not need to call this. It is considered + * an advanced technique and is mostly used for testing. + * + * 

+     * int64_t timeoutNanos = 500 * kNanosPerMillisecond; // arbitrary 1/2 second
+     * StreamState currentState = stream->getState();
+     * StreamState nextState = StreamState::Unknown;
+     * while (result == Result::OK && currentState != StreamState::Paused) {
+     *     result = stream->waitForStateChange(
+     *                                   currentState, &nextState, timeoutNanos);
+     *     currentState = nextState;
+     * }
+     *
+ * + * If the state does not change within the timeout period then it will + * return ErrorTimeout. This is true even if timeoutNanoseconds is zero. + * + * @param inputState The state we want to change away from. + * @param nextState Pointer to a variable that will be set to the new state. + * @param timeoutNanoseconds The maximum time to wait in nanoseconds. + * @return Result::OK or a Result::Error. + */ + virtual Result waitForStateChange(StreamState inputState, + StreamState *nextState, + int64_t timeoutNanoseconds) = 0; + + /** + * This can be used to adjust the latency of the buffer by changing + * the threshold where blocking will occur. + * By combining this with getXRunCount(), the latency can be tuned + * at run-time for each device. + * + * This cannot be set higher than getBufferCapacity(). + * + * @param requestedFrames requested number of frames that can be filled without blocking + * @return the resulting buffer size in frames (obtained using value()) or an error (obtained + * using error()) + */ + virtual ResultWithValue setBufferSizeInFrames(int32_t /* requestedFrames */) { + return Result::ErrorUnimplemented; + } + + /** + * An XRun is an Underrun or an Overrun. + * During playing, an underrun will occur if the stream is not written in time + * and the system runs out of valid data. + * During recording, an overrun will occur if the stream is not read in time + * and there is no place to put the incoming data so it is discarded. + * + * An underrun or overrun can cause an audible "pop" or "glitch". + * + * @return a result which is either Result::OK with the xRun count as the value, or a + * Result::Error* code + */ + virtual ResultWithValue getXRunCount() const { + return ResultWithValue(Result::ErrorUnimplemented); + } + + /** + * @return true if XRun counts are supported on the stream + */ + virtual bool isXRunCountSupported() const = 0; + + /** + * Query the number of frames that are read or written by the endpoint at one time. + * + * @return burst size + */ + virtual int32_t getFramesPerBurst() = 0; + + /** + * Get the number of bytes in each audio frame. This is calculated using the channel count + * and the sample format. For example, a 2 channel floating point stream will have + * 2 * 4 = 8 bytes per frame. + * + * @return number of bytes in each audio frame. + */ + int32_t getBytesPerFrame() const { return mChannelCount * getBytesPerSample(); } + + /** + * Get the number of bytes per sample. This is calculated using the sample format. For example, + * a stream using 16-bit integer samples will have 2 bytes per sample. + * + * @return the number of bytes per sample. + */ + int32_t getBytesPerSample() const; + + /** + * The number of audio frames written into the stream. + * This monotonic counter will never get reset. + * + * @return the number of frames written so far + */ + virtual int64_t getFramesWritten(); + + /** + * The number of audio frames read from the stream. + * This monotonic counter will never get reset. + * + * @return the number of frames read so far + */ + virtual int64_t getFramesRead(); + + /** + * Calculate the latency of a stream based on getTimestamp(). + * + * Output latency is the time it takes for a given frame to travel from the + * app to some type of digital-to-analog converter. If the DAC is external, for example + * in a USB interface or a TV connected by HDMI, then there may be additional latency + * that the Android device is unaware of. + * + * Input latency is the time it takes to a given frame to travel from an analog-to-digital + * converter (ADC) to the app. + * + * Note that the latency of an OUTPUT stream will increase abruptly when you write data to it + * and then decrease slowly over time as the data is consumed. + * + * The latency of an INPUT stream will decrease abruptly when you read data from it + * and then increase slowly over time as more data arrives. + * + * The latency of an OUTPUT stream is generally higher than the INPUT latency + * because an app generally tries to keep the OUTPUT buffer full and the INPUT buffer empty. + * + * @return a ResultWithValue which has a result of Result::OK and a value containing the latency + * in milliseconds, or a result of Result::Error*. + */ + virtual ResultWithValue calculateLatencyMillis() { + return ResultWithValue(Result::ErrorUnimplemented); + } + + /** + * Get the estimated time that the frame at `framePosition` entered or left the audio processing + * pipeline. + * + * This can be used to coordinate events and interactions with the external environment, and to + * estimate the latency of an audio stream. An example of usage can be found in the hello-oboe + * sample (search for "calculateCurrentOutputLatencyMillis"). + * + * The time is based on the implementation's best effort, using whatever knowledge is available + * to the system, but cannot account for any delay unknown to the implementation. + * + * @deprecated since 1.0, use AudioStream::getTimestamp(clockid_t clockId) instead, which + * returns ResultWithValue + * @param clockId the type of clock to use e.g. CLOCK_MONOTONIC + * @param framePosition the frame number to query + * @param timeNanoseconds an output parameter which will contain the presentation timestamp + */ + virtual Result getTimestamp(clockid_t /* clockId */, + int64_t* /* framePosition */, + int64_t* /* timeNanoseconds */) { + return Result::ErrorUnimplemented; + } + + /** + * Get the estimated time that the frame at `framePosition` entered or left the audio processing + * pipeline. + * + * This can be used to coordinate events and interactions with the external environment, and to + * estimate the latency of an audio stream. An example of usage can be found in the hello-oboe + * sample (search for "calculateCurrentOutputLatencyMillis"). + * + * The time is based on the implementation's best effort, using whatever knowledge is available + * to the system, but cannot account for any delay unknown to the implementation. + * + * @param clockId the type of clock to use e.g. CLOCK_MONOTONIC + * @return a FrameTimestamp containing the position and time at which a particular audio frame + * entered or left the audio processing pipeline, or an error if the operation failed. + */ + virtual ResultWithValue getTimestamp(clockid_t /* clockId */); + + // ============== I/O =========================== + /** + * Write data from the supplied buffer into the stream. This method will block until the write + * is complete or it runs out of time. + * + * If `timeoutNanoseconds` is zero then this call will not wait. + * + * @param buffer The address of the first sample. + * @param numFrames Number of frames to write. Only complete frames will be written. + * @param timeoutNanoseconds Maximum number of nanoseconds to wait for completion. + * @return a ResultWithValue which has a result of Result::OK and a value containing the number + * of frames actually written, or result of Result::Error*. + */ + virtual ResultWithValue write(const void* /* buffer */, + int32_t /* numFrames */, + int64_t /* timeoutNanoseconds */ ) { + return ResultWithValue(Result::ErrorUnimplemented); + } + + /** + * Read data into the supplied buffer from the stream. This method will block until the read + * is complete or it runs out of time. + * + * If `timeoutNanoseconds` is zero then this call will not wait. + * + * @param buffer The address of the first sample. + * @param numFrames Number of frames to read. Only complete frames will be read. + * @param timeoutNanoseconds Maximum number of nanoseconds to wait for completion. + * @return a ResultWithValue which has a result of Result::OK and a value containing the number + * of frames actually read, or result of Result::Error*. + */ + virtual ResultWithValue read(void* /* buffer */, + int32_t /* numFrames */, + int64_t /* timeoutNanoseconds */) { + return ResultWithValue(Result::ErrorUnimplemented); + } + + /** + * Get the underlying audio API which the stream uses. + * + * @return the API that this stream uses. + */ + virtual AudioApi getAudioApi() const = 0; + + /** + * Returns true if the underlying audio API is AAudio. + * + * @return true if this stream is implemented using the AAudio API. + */ + bool usesAAudio() const { + return getAudioApi() == AudioApi::AAudio; + } + + /** + * Only for debugging. Do not use in production. + * If you need to call this method something is wrong. + * If you think you need it for production then please let us know + * so we can modify Oboe so that you don't need this. + * + * @return nullptr or a pointer to a stream from the system API + */ + virtual void *getUnderlyingStream() const { + return nullptr; + } + + /** + * Launch a thread that will stop the stream. + */ + void launchStopThread(); + + /** + * Update mFramesWritten. + * For internal use only. + */ + virtual void updateFramesWritten() = 0; + + /** + * Update mFramesRead. + * For internal use only. + */ + virtual void updateFramesRead() = 0; + + /* + * Swap old callback for new callback. + * This not atomic. + * This should only be used internally. + * @param streamCallback + * @return previous streamCallback + */ + AudioStreamCallback *swapCallback(AudioStreamCallback *streamCallback) { + AudioStreamCallback *previousCallback = mStreamCallback; + mStreamCallback = streamCallback; + return previousCallback; + } + + /** + * @return number of frames of data currently in the buffer + */ + ResultWithValue getAvailableFrames(); + + /** + * Wait until the stream has a minimum amount of data available in its buffer. + * This can be used with an EXCLUSIVE MMAP input stream to avoid reading data too close to + * the DSP write position, which may cause glitches. + * + * @param numFrames minimum frames available + * @param timeoutNanoseconds + * @return number of frames available, ErrorTimeout + */ + ResultWithValue waitForAvailableFrames(int32_t numFrames, + int64_t timeoutNanoseconds); + +protected: + + /** + * This is used to detect more than one error callback from a stream. + * These were bugs in some versions of Android that caused multiple error callbacks. + * Internal bug b/63087953 + * + * Calling this sets an atomic true and returns the previous value. + * + * @return false on first call, true on subsequent calls + */ + bool wasErrorCallbackCalled() { + return mErrorCallbackCalled.exchange(true); + } + + /** + * Wait for a transition from one state to another. + * @return OK if the endingState was observed, or ErrorUnexpectedState + * if any state that was not the startingState or endingState was observed + * or ErrorTimeout. + */ + virtual Result waitForStateTransition(StreamState startingState, + StreamState endingState, + int64_t timeoutNanoseconds); + + /** + * Override this to provide a default for when the application did not specify a callback. + * + * @param audioData + * @param numFrames + * @return result + */ + virtual DataCallbackResult onDefaultCallback(void* /* audioData */, int /* numFrames */) { + return DataCallbackResult::Stop; + } + + /** + * Override this to provide your own behaviour for the audio callback + * + * @param audioData container array which audio frames will be written into or read from + * @param numFrames number of frames which were read/written + * @return the result of the callback: stop or continue + * + */ + DataCallbackResult fireDataCallback(void *audioData, int numFrames); + + /** + * @return true if callbacks may be called + */ + bool isDataCallbackEnabled() { + return mDataCallbackEnabled; + } + + /** + * This can be set false internally to prevent callbacks + * after DataCallbackResult::Stop has been returned. + */ + void setDataCallbackEnabled(bool enabled) { + mDataCallbackEnabled = enabled; + } + + /** + * Number of frames which have been written into the stream + * + * This is signed integer to match the counters in AAudio. + * At audio rates, the counter will overflow in about six million years. + */ + std::atomic mFramesWritten{}; + + /** + * Number of frames which have been read from the stream. + * + * This is signed integer to match the counters in AAudio. + * At audio rates, the counter will overflow in about six million years. + */ + std::atomic mFramesRead{}; + + std::mutex mLock; // for synchronizing start/stop/close + +private: + int mPreviousScheduler = -1; + + std::atomic mDataCallbackEnabled{false}; + std::atomic mErrorCallbackCalled{false}; + + +}; + +/** + * This struct is a stateless functor which closes a audiostream prior to its deletion. + * This means it can be used to safely delete a smart pointer referring to an open stream. + */ + struct StreamDeleterFunctor { + void operator()(AudioStream *audioStream) { + if (audioStream) { + audioStream->close(); + } + delete audioStream; + } + }; +} // namespace oboe + +#endif /* OBOE_STREAM_H_ */ diff --git a/modules/juce_audio_devices/native/oboe/include/oboe/AudioStreamBase.h b/modules/juce_audio_devices/native/oboe/include/oboe/AudioStreamBase.h new file mode 100644 index 0000000000..fd327e6e87 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/include/oboe/AudioStreamBase.h @@ -0,0 +1,201 @@ +/* + * Copyright 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_STREAM_BASE_H_ +#define OBOE_STREAM_BASE_H_ + +#include +#include "oboe/AudioStreamCallback.h" +#include "oboe/Definitions.h" + +namespace oboe { + +/** + * Base class containing parameters for audio streams and builders. + **/ +class AudioStreamBase { +public: + + AudioStreamBase() {} + + virtual ~AudioStreamBase() = default; + + // This class only contains primitives so we can use default constructor and copy methods. + + /** + * Default copy constructor + */ + AudioStreamBase(const AudioStreamBase&) = default; + + /** + * Default assignment operator + */ + AudioStreamBase& operator=(const AudioStreamBase&) = default; + + /** + * @return number of channels, for example 2 for stereo, or kUnspecified + */ + int32_t getChannelCount() const { return mChannelCount; } + + /** + * @return Direction::Input or Direction::Output + */ + Direction getDirection() const { return mDirection; } + + /** + * @return sample rate for the stream or kUnspecified + */ + int32_t getSampleRate() const { return mSampleRate; } + + /** + * @return the number of frames in each callback or kUnspecified. + */ + int32_t getFramesPerCallback() const { return mFramesPerCallback; } + + /** + * @return the audio sample format (e.g. Float or I16) + */ + AudioFormat getFormat() const { return mFormat; } + + /** + * Query the maximum number of frames that can be filled without blocking. + * If the stream has been closed the last known value will be returned. + * + * @return buffer size + */ + virtual int32_t getBufferSizeInFrames() { return mBufferSizeInFrames; } + + /** + * @return capacityInFrames or kUnspecified + */ + virtual int32_t getBufferCapacityInFrames() const { return mBufferCapacityInFrames; } + + /** + * @return the sharing mode of the stream. + */ + SharingMode getSharingMode() const { return mSharingMode; } + + /** + * @return the performance mode of the stream. + */ + PerformanceMode getPerformanceMode() const { return mPerformanceMode; } + + /** + * @return the device ID of the stream. + */ + int32_t getDeviceId() const { return mDeviceId; } + + /** + * @return the callback object for this stream, if set. + */ + AudioStreamCallback* getCallback() const { + return mStreamCallback; + } + + /** + * @return the usage for this stream. + */ + Usage getUsage() const { return mUsage; } + + /** + * @return the stream's content type. + */ + ContentType getContentType() const { return mContentType; } + + /** + * @return the stream's input preset. + */ + InputPreset getInputPreset() const { return mInputPreset; } + + /** + * @return the stream's session ID allocation strategy (None or Allocate). + */ + SessionId getSessionId() const { return mSessionId; } + + /** + * @return true if Oboe can convert channel counts to achieve optimal results. + */ + bool isChannelConversionAllowed() const { + return mChannelConversionAllowed; + } + + /** + * @return true if Oboe can convert data formats to achieve optimal results. + */ + bool isFormatConversionAllowed() const { + return mFormatConversionAllowed; + } + + /** + * @return whether and how Oboe can convert sample rates to achieve optimal results. + */ + SampleRateConversionQuality getSampleRateConversionQuality() const { + return mSampleRateConversionQuality; + } + +protected: + + /** The callback which will be fired when new data is ready to be read/written **/ + AudioStreamCallback *mStreamCallback = nullptr; + /** Number of audio frames which will be requested in each callback */ + int32_t mFramesPerCallback = kUnspecified; + /** Stream channel count */ + int32_t mChannelCount = kUnspecified; + /** Stream sample rate */ + int32_t mSampleRate = kUnspecified; + /** Stream audio device ID */ + int32_t mDeviceId = kUnspecified; + /** Stream buffer capacity specified as a number of audio frames */ + int32_t mBufferCapacityInFrames = kUnspecified; + /** Stream buffer size specified as a number of audio frames */ + int32_t mBufferSizeInFrames = kUnspecified; + /** + * Number of frames which will be copied to/from the audio device in a single read/write + * operation + */ + int32_t mFramesPerBurst = kUnspecified; + + /** Stream sharing mode */ + SharingMode mSharingMode = SharingMode::Shared; + /** Format of audio frames */ + AudioFormat mFormat = AudioFormat::Unspecified; + /** Stream direction */ + Direction mDirection = Direction::Output; + /** Stream performance mode */ + PerformanceMode mPerformanceMode = PerformanceMode::None; + + /** Stream usage. Only active on Android 28+ */ + Usage mUsage = Usage::Media; + /** Stream content type. Only active on Android 28+ */ + ContentType mContentType = ContentType::Music; + /** Stream input preset. Only active on Android 28+ + * TODO InputPreset::Unspecified should be considered as a possible default alternative. + */ + InputPreset mInputPreset = InputPreset::VoiceRecognition; + /** Stream session ID allocation strategy. Only active on Android 28+ */ + SessionId mSessionId = SessionId::None; + + // Control whether Oboe can convert channel counts to achieve optimal results. + bool mChannelConversionAllowed = false; + // Control whether Oboe can convert data formats to achieve optimal results. + bool mFormatConversionAllowed = false; + // Control whether and how Oboe can convert sample rates to achieve optimal results. + SampleRateConversionQuality mSampleRateConversionQuality = SampleRateConversionQuality::None; +}; + +} // namespace oboe + +#endif /* OBOE_STREAM_BASE_H_ */ diff --git a/modules/juce_audio_devices/native/oboe/include/oboe/AudioStreamBuilder.h b/modules/juce_audio_devices/native/oboe/include/oboe/AudioStreamBuilder.h new file mode 100644 index 0000000000..b0f7de8e7c --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/include/oboe/AudioStreamBuilder.h @@ -0,0 +1,424 @@ +/* + * Copyright 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_STREAM_BUILDER_H_ +#define OBOE_STREAM_BUILDER_H_ + +#include "oboe/Definitions.h" +#include "oboe/AudioStreamBase.h" + +namespace oboe { + + // This depends on AudioStream, so we use forward declaration, it will close and delete the stream + struct StreamDeleterFunctor; + using ManagedStream = std::unique_ptr; +/** + * Factory class for an audio Stream. + */ +class AudioStreamBuilder : public AudioStreamBase { +public: + + AudioStreamBuilder() : AudioStreamBase() {} + + AudioStreamBuilder(const AudioStreamBase &audioStreamBase): AudioStreamBase(audioStreamBase) {} + + /** + * Request a specific number of channels. + * + * Default is kUnspecified. If the value is unspecified then + * the application should query for the actual value after the stream is opened. + */ + AudioStreamBuilder *setChannelCount(int channelCount) { + mChannelCount = channelCount; + return this; + } + + /** + * Request the direction for a stream. The default is Direction::Output. + * + * @param direction Direction::Output or Direction::Input + */ + AudioStreamBuilder *setDirection(Direction direction) { + mDirection = direction; + return this; + } + + /** + * Request a specific sample rate in Hz. + * + * Default is kUnspecified. If the value is unspecified then + * the application should query for the actual value after the stream is opened. + * + * Technically, this should be called the "frame rate" or "frames per second", + * because it refers to the number of complete frames transferred per second. + * But it is traditionally called "sample rate". Se we use that term. + * + */ + AudioStreamBuilder *setSampleRate(int32_t sampleRate) { + mSampleRate = sampleRate; + return this; + } + + /** + * Request a specific number of frames for the data callback. + * + * Default is kUnspecified. If the value is unspecified then + * the actual number may vary from callback to callback. + * + * If an application can handle a varying number of frames then we recommend + * leaving this unspecified. This allow the underlying API to optimize + * the callbacks. But if your application is, for example, doing FFTs or other block + * oriented operations, then call this function to get the sizes you need. + * + * @param framesPerCallback + * @return pointer to the builder so calls can be chained + */ + AudioStreamBuilder *setFramesPerCallback(int framesPerCallback) { + mFramesPerCallback = framesPerCallback; + return this; + } + + /** + * Request a sample data format, for example Format::Float. + * + * Default is Format::Unspecified. If the value is unspecified then + * the application should query for the actual value after the stream is opened. + */ + AudioStreamBuilder *setFormat(AudioFormat format) { + mFormat = format; + return this; + } + + /** + * Set the requested buffer capacity in frames. + * BufferCapacityInFrames is the maximum possible BufferSizeInFrames. + * + * The final stream capacity may differ. For AAudio it should be at least this big. + * For OpenSL ES, it could be smaller. + * + * Default is kUnspecified. + * + * @param bufferCapacityInFrames the desired buffer capacity in frames or kUnspecified + * @return pointer to the builder so calls can be chained + */ + AudioStreamBuilder *setBufferCapacityInFrames(int32_t bufferCapacityInFrames) { + mBufferCapacityInFrames = bufferCapacityInFrames; + return this; + } + + /** + * Get the audio API which will be requested when opening the stream. No guarantees that this is + * the API which will actually be used. Query the stream itself to find out the API which is + * being used. + * + * If you do not specify the API, then AAudio will be used if isAAudioRecommended() + * returns true. Otherwise OpenSL ES will be used. + * + * @return the requested audio API + */ + AudioApi getAudioApi() const { return mAudioApi; } + + /** + * If you leave this unspecified then Oboe will choose the best API + * for the device and SDK version at runtime. + * + * This should almost always be left unspecified, except for debugging purposes. + * Specifying AAudio will force Oboe to use AAudio on 8.0, which is extremely risky. + * Specifying OpenSLES should mainly be used to test legacy performance/functionality. + * + * If the caller requests AAudio and it is supported then AAudio will be used. + * + * @param audioApi Must be AudioApi::Unspecified, AudioApi::OpenSLES or AudioApi::AAudio. + * @return pointer to the builder so calls can be chained + */ + AudioStreamBuilder *setAudioApi(AudioApi audioApi) { + mAudioApi = audioApi; + return this; + } + + /** + * Is the AAudio API supported on this device? + * + * AAudio was introduced in the Oreo 8.0 release. + * + * @return true if supported + */ + static bool isAAudioSupported(); + + /** + * Is the AAudio API recommended this device? + * + * AAudio may be supported but not recommended because of version specific issues. + * AAudio is not recommended for Android 8.0 or earlier versions. + * + * @return true if recommended + */ + static bool isAAudioRecommended(); + + /** + * Request a mode for sharing the device. + * The requested sharing mode may not be available. + * So the application should query for the actual mode after the stream is opened. + * + * @param sharingMode SharingMode::Shared or SharingMode::Exclusive + * @return pointer to the builder so calls can be chained + */ + AudioStreamBuilder *setSharingMode(SharingMode sharingMode) { + mSharingMode = sharingMode; + return this; + } + + /** + * Request a performance level for the stream. + * This will determine the latency, the power consumption, and the level of + * protection from glitches. + * + * @param performanceMode for example, PerformanceMode::LowLatency + * @return pointer to the builder so calls can be chained + */ + AudioStreamBuilder *setPerformanceMode(PerformanceMode performanceMode) { + mPerformanceMode = performanceMode; + return this; + } + + + /** + * Set the intended use case for the stream. + * + * The system will use this information to optimize the behavior of the stream. + * This could, for example, affect how volume and focus is handled for the stream. + * + * The default, if you do not call this function, is Usage::Media. + * + * Added in API level 28. + * + * @param usage the desired usage, eg. Usage::Game + */ + AudioStreamBuilder *setUsage(Usage usage) { + mUsage = usage; + return this; + } + + /** + * Set the type of audio data that the stream will carry. + * + * The system will use this information to optimize the behavior of the stream. + * This could, for example, affect whether a stream is paused when a notification occurs. + * + * The default, if you do not call this function, is ContentType::Music. + * + * Added in API level 28. + * + * @param contentType the type of audio data, eg. ContentType::Speech + */ + AudioStreamBuilder *setContentType(ContentType contentType) { + mContentType = contentType; + return this; + } + + /** + * Set the input (capture) preset for the stream. + * + * The system will use this information to optimize the behavior of the stream. + * This could, for example, affect which microphones are used and how the + * recorded data is processed. + * + * The default, if you do not call this function, is InputPreset::VoiceRecognition. + * That is because VoiceRecognition is the preset with the lowest latency + * on many platforms. + * + * Added in API level 28. + * + * @param inputPreset the desired configuration for recording + */ + AudioStreamBuilder *setInputPreset(InputPreset inputPreset) { + mInputPreset = inputPreset; + return this; + } + + /** Set the requested session ID. + * + * The session ID can be used to associate a stream with effects processors. + * The effects are controlled using the Android AudioEffect Java API. + * + * The default, if you do not call this function, is SessionId::None. + * + * If set to SessionId::Allocate then a session ID will be allocated + * when the stream is opened. + * + * The allocated session ID can be obtained by calling AudioStream::getSessionId() + * and then used with this function when opening another stream. + * This allows effects to be shared between streams. + * + * Session IDs from Oboe can be used the Android Java APIs and vice versa. + * So a session ID from an Oboe stream can be passed to Java + * and effects applied using the Java AudioEffect API. + * + * Allocated session IDs will always be positive and nonzero. + * + * Added in API level 28. + * + * @param sessionId an allocated sessionID or SessionId::Allocate + */ + AudioStreamBuilder *setSessionId(SessionId sessionId) { + mSessionId = sessionId; + return this; + } + + /** + * Request a stream to a specific audio input/output device given an audio device ID. + * + * In most cases, the primary device will be the appropriate device to use, and the + * deviceId can be left kUnspecified. + * + * On Android, for example, the ID could be obtained from the Java AudioManager. + * AudioManager.getDevices() returns an array of AudioDeviceInfo[], which contains + * a getId() method (as well as other type information), that should be passed + * to this method. + * + * + * Note that when using OpenSL ES, this will be ignored and the created + * stream will have deviceId kUnspecified. + * + * @param deviceId device identifier or kUnspecified + * @return pointer to the builder so calls can be chained + */ + AudioStreamBuilder *setDeviceId(int32_t deviceId) { + mDeviceId = deviceId; + return this; + } + + /** + * Specifies an object to handle data or error related callbacks from the underlying API. + * + * Important: See AudioStreamCallback for restrictions on what may be called + * from the callback methods. + * + * When an error callback occurs, the associated stream will be stopped and closed in a separate thread. + * + * A note on why the streamCallback parameter is a raw pointer rather than a smart pointer: + * + * The caller should retain ownership of the object streamCallback points to. At first glance weak_ptr may seem like + * a good candidate for streamCallback as this implies temporary ownership. However, a weak_ptr can only be created + * from a shared_ptr. A shared_ptr incurs some performance overhead. The callback object is likely to be accessed + * every few milliseconds when the stream requires new data so this overhead is something we want to avoid. + * + * This leaves a raw pointer as the logical type choice. The only caveat being that the caller must not destroy + * the callback before the stream has been closed. + * + * @param streamCallback + * @return pointer to the builder so calls can be chained + */ + AudioStreamBuilder *setCallback(AudioStreamCallback *streamCallback) { + mStreamCallback = streamCallback; + return this; + } + + /** + * If true then Oboe might convert channel counts to achieve optimal results. + * On some versions of Android for example, stereo streams could not use a FAST track. + * So a mono stream might be used instead and duplicated to two channels. + * On some devices, mono streams might be broken, so a stereo stream might be opened + * and converted to mono. + * + * Default is true. + */ + AudioStreamBuilder *setChannelConversionAllowed(bool allowed) { + mChannelConversionAllowed = allowed; + return this; + } + + /** + * If true then Oboe might convert data formats to achieve optimal results. + * On some versions of Android, for example, a float stream could not get a + * low latency data path. So an I16 stream might be opened and converted to float. + * + * Default is true. + */ + AudioStreamBuilder *setFormatConversionAllowed(bool allowed) { + mFormatConversionAllowed = allowed; + return this; + } + + /** + * Specify the quality of the sample rate converter in Oboe. + * + * If set to None then Oboe will not do sample rate conversion. But the underlying APIs might + * still do sample rate conversion if you specify a sample rate. + * That can prevent you from getting a low latency stream. + * + * If you do the conversion in Oboe then you might still get a low latency stream. + * + * Default is SampleRateConversionQuality::None + */ + AudioStreamBuilder *setSampleRateConversionQuality(SampleRateConversionQuality quality) { + mSampleRateConversionQuality = quality; + return this; + } + + /** + * @return true if AAudio will be used based on the current settings. + */ + bool willUseAAudio() const { + return (mAudioApi == AudioApi::AAudio && isAAudioSupported()) + || (mAudioApi == AudioApi::Unspecified && isAAudioRecommended()); + } + + /** + * Create and open a stream object based on the current settings. + * + * The caller owns the pointer to the AudioStream object. + * + * @param stream pointer to a variable to receive the stream address + * @return OBOE_OK if successful or a negative error code + */ + Result openStream(AudioStream **stream); + + + /** + * Create and open a ManagedStream object based on the current builder state. + * + * The caller must create a unique ptr, and pass by reference so it can be + * modified to point to an opened stream. The caller owns the unique ptr, + * and it will be automatically closed and deleted when going out of scope. + * @param stream Reference to the ManagedStream (uniqueptr) used to keep track of stream + * @return OBOE_OK if successful or a negative error code. + */ + Result openManagedStream(ManagedStream &stream); + +private: + + /** + * @param other + * @return true if channels, format and sample rate match + */ + bool isCompatible(AudioStreamBase &other); + + /** + * Create an AudioStream object. The AudioStream must be opened before use. + * + * The caller owns the pointer. + * + * @return pointer to an AudioStream object or nullptr. + */ + oboe::AudioStream *build(); + + AudioApi mAudioApi = AudioApi::Unspecified; +}; + +} // namespace oboe + +#endif /* OBOE_STREAM_BUILDER_H_ */ diff --git a/modules/juce_audio_devices/native/oboe/include/oboe/AudioStreamCallback.h b/modules/juce_audio_devices/native/oboe/include/oboe/AudioStreamCallback.h new file mode 100644 index 0000000000..f427693e30 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/include/oboe/AudioStreamCallback.h @@ -0,0 +1,123 @@ +/* + * Copyright (C) 2016 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_STREAM_CALLBACK_H +#define OBOE_STREAM_CALLBACK_H + +#include "oboe/Definitions.h" + +namespace oboe { + +class AudioStream; + +/** + * AudioStreamCallback defines a callback interface for: + * + * 1) moving data to/from an audio stream using `onAudioReady` + * 2) being alerted when a stream has an error using `onError*` methods + * + */ +class AudioStreamCallback { +public: + virtual ~AudioStreamCallback() = default; + + /** + * A buffer is ready for processing. + * + * For an output stream, this function should render and write numFrames of data + * in the stream's current data format to the audioData buffer. + * + * For an input stream, this function should read and process numFrames of data + * from the audioData buffer. + * + * The audio data is passed through the buffer. So do NOT call read() or + * write() on the stream that is making the callback. + * + * Note that numFrames can vary unless AudioStreamBuilder::setFramesPerCallback() + * is called. + * + * Also note that this callback function should be considered a "real-time" function. + * It must not do anything that could cause an unbounded delay because that can cause the + * audio to glitch or pop. + * + * These are things the function should NOT do: + *
    + *
  • allocate memory using, for example, malloc() or new
    • + *
  • any file operations such as opening, closing, reading or writing
    • + *
  • any network operations such as streaming
    • + *
  • use any mutexes or other synchronization primitives
    • + *
  • sleep
    • + *
  • oboeStream->stop(), pause(), flush() or close()
    • + *
  • oboeStream->read()
    • + *
  • oboeStream->write()
    • + *
+ * + * The following are OK to call from the data callback: + *
    + *
  • oboeStream->get*()
    • + *
  • oboe::convertToText()
    • + *
  • oboeStream->setBufferSizeInFrames()
    • + *
+ * + * If you need to move data, eg. MIDI commands, in or out of the callback function then + * we recommend the use of non-blocking techniques such as an atomic FIFO. + * + * @param oboeStream pointer to the associated stream + * @param audioData buffer containing input data or a place to put output data + * @param numFrames number of frames to be processed + * @return DataCallbackResult::Continue or DataCallbackResult::Stop + */ + virtual DataCallbackResult onAudioReady( + AudioStream *oboeStream, + void *audioData, + int32_t numFrames) = 0; + + /** + * This will be called when an error occurs on a stream or when the stream is disconnected. + * + * Note that this will be called on a different thread than the onAudioReady() thread. + * This thread will be created by Oboe. + * + * The underlying stream will already be stopped by Oboe but not yet closed. + * So the stream can be queried. + * + * Do not close or delete the stream in this method because it will be + * closed after this method returns. + * + * @param oboeStream pointer to the associated stream + * @param error + */ + virtual void onErrorBeforeClose(AudioStream* /* oboeStream */, Result /* error */) {} + + /** + * This will be called when an error occurs on a stream or when the stream is disconnected. + * The underlying AAudio or OpenSL ES stream will already be stopped AND closed by Oboe. + * So the underlying stream cannot be referenced. + * But you can still query most parameters. + * + * This callback could be used to reopen a new stream on another device. + * You can safely delete the old AudioStream in this method. + * + * @param oboeStream pointer to the associated stream + * @param error + */ + virtual void onErrorAfterClose(AudioStream* /* oboeStream */, Result /* error */) {} + +}; + +} // namespace oboe + +#endif //OBOE_STREAM_CALLBACK_H diff --git a/modules/juce_audio_devices/native/oboe/include/oboe/Definitions.h b/modules/juce_audio_devices/native/oboe/include/oboe/Definitions.h new file mode 100644 index 0000000000..04c1b68286 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/include/oboe/Definitions.h @@ -0,0 +1,500 @@ +/* + * Copyright (C) 2016 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_DEFINITIONS_H +#define OBOE_DEFINITIONS_H + + +#include +#include + +// Oboe needs to be able to build on old NDKs so we use hard coded constants. +// The correctness of these constants is verified in "aaudio/AAudioLoader.cpp". + +namespace oboe { + + /** + * Represents any attribute, property or value which hasn't been specified. + */ + constexpr int32_t kUnspecified = 0; + + // TODO: Investigate using std::chrono + /** + * The number of nanoseconds in a microsecond. 1,000. + */ + constexpr int64_t kNanosPerMicrosecond = 1000; + + /** + * The number of nanoseconds in a millisecond. 1,000,000. + */ + constexpr int64_t kNanosPerMillisecond = kNanosPerMicrosecond * 1000; + + /** + * The number of milliseconds in a second. 1,000. + */ + constexpr int64_t kMillisPerSecond = 1000; + + /** + * The number of nanoseconds in a second. 1,000,000,000. + */ + constexpr int64_t kNanosPerSecond = kNanosPerMillisecond * kMillisPerSecond; + + /** + * The state of the audio stream. + */ + enum class StreamState : int32_t { // aaudio_stream_state_t + Uninitialized = 0, // AAUDIO_STREAM_STATE_UNINITIALIZED, + Unknown = 1, // AAUDIO_STREAM_STATE_UNKNOWN, + Open = 2, // AAUDIO_STREAM_STATE_OPEN, + Starting = 3, // AAUDIO_STREAM_STATE_STARTING, + Started = 4, // AAUDIO_STREAM_STATE_STARTED, + Pausing = 5, // AAUDIO_STREAM_STATE_PAUSING, + Paused = 6, // AAUDIO_STREAM_STATE_PAUSED, + Flushing = 7, // AAUDIO_STREAM_STATE_FLUSHING, + Flushed = 8, // AAUDIO_STREAM_STATE_FLUSHED, + Stopping = 9, // AAUDIO_STREAM_STATE_STOPPING, + Stopped = 10, // AAUDIO_STREAM_STATE_STOPPED, + Closing = 11, // AAUDIO_STREAM_STATE_CLOSING, + Closed = 12, // AAUDIO_STREAM_STATE_CLOSED, + Disconnected = 13, // AAUDIO_STREAM_STATE_DISCONNECTED, + }; + + /** + * The direction of the stream. + */ + enum class Direction : int32_t { // aaudio_direction_t + + /** + * Used for playback. + */ + Output = 0, // AAUDIO_DIRECTION_OUTPUT, + + /** + * Used for recording. + */ + Input = 1, // AAUDIO_DIRECTION_INPUT, + }; + + /** + * The format of audio samples. + */ + enum class AudioFormat : int32_t { // aaudio_format_t + /** + * Invalid format. + */ + Invalid = -1, // AAUDIO_FORMAT_INVALID, + + /** + * Unspecified format. Format will be decided by Oboe. + */ + Unspecified = 0, // AAUDIO_FORMAT_UNSPECIFIED, + + /** + * Signed 16-bit integers. + */ + I16 = 1, // AAUDIO_FORMAT_PCM_I16, + + /** + * Single precision floating points. + */ + Float = 2, // AAUDIO_FORMAT_PCM_FLOAT, + }; + + /** + * The result of an audio callback. + */ + enum class DataCallbackResult : int32_t { // aaudio_data_callback_result_t + // Indicates to the caller that the callbacks should continue. + Continue = 0, // AAUDIO_CALLBACK_RESULT_CONTINUE, + + // Indicates to the caller that the callbacks should stop immediately. + Stop = 1, // AAUDIO_CALLBACK_RESULT_STOP, + }; + + /** + * The result of an operation. All except the `OK` result indicates that an error occurred. + * The `Result` can be converted into a human readable string using `convertToText`. + */ + enum class Result : int32_t { // aaudio_result_t + OK = 0, // AAUDIO_OK + ErrorBase = -900, // AAUDIO_ERROR_BASE, + ErrorDisconnected = -899, // AAUDIO_ERROR_DISCONNECTED, + ErrorIllegalArgument = -898, // AAUDIO_ERROR_ILLEGAL_ARGUMENT, + ErrorInternal = -896, // AAUDIO_ERROR_INTERNAL, + ErrorInvalidState = -895, // AAUDIO_ERROR_INVALID_STATE, + ErrorInvalidHandle = -892, // AAUDIO_ERROR_INVALID_HANDLE, + ErrorUnimplemented = -890, // AAUDIO_ERROR_UNIMPLEMENTED, + ErrorUnavailable = -889, // AAUDIO_ERROR_UNAVAILABLE, + ErrorNoFreeHandles = -888, // AAUDIO_ERROR_NO_FREE_HANDLES, + ErrorNoMemory = -887, // AAUDIO_ERROR_NO_MEMORY, + ErrorNull = -886, // AAUDIO_ERROR_NULL, + ErrorTimeout = -885, // AAUDIO_ERROR_TIMEOUT, + ErrorWouldBlock = -884, // AAUDIO_ERROR_WOULD_BLOCK, + ErrorInvalidFormat = -883, // AAUDIO_ERROR_INVALID_FORMAT, + ErrorOutOfRange = -882, // AAUDIO_ERROR_OUT_OF_RANGE, + ErrorNoService = -881, // AAUDIO_ERROR_NO_SERVICE, + ErrorInvalidRate = -880, // AAUDIO_ERROR_INVALID_RATE, + // Reserved for future AAudio result types + Reserved1, + Reserved2, + Reserved3, + Reserved4, + Reserved5, + Reserved6, + Reserved7, + Reserved8, + Reserved9, + Reserved10, + ErrorClosed, + }; + + /** + * The sharing mode of the audio stream. + */ + enum class SharingMode : int32_t { // aaudio_sharing_mode_t + + /** + * This will be the only stream using a particular source or sink. + * This mode will provide the lowest possible latency. + * You should close EXCLUSIVE streams immediately when you are not using them. + * + * If you do not need the lowest possible latency then we recommend using Shared, + * which is the default. + */ + Exclusive = 0, // AAUDIO_SHARING_MODE_EXCLUSIVE, + + /** + * Multiple applications can share the same device. + * The data from output streams will be mixed by the audio service. + * The data for input streams will be distributed by the audio service. + * + * This will have higher latency than the EXCLUSIVE mode. + */ + Shared = 1, // AAUDIO_SHARING_MODE_SHARED, + }; + + /** + * The performance mode of the audio stream. + */ + enum class PerformanceMode : int32_t { // aaudio_performance_mode_t + + /** + * No particular performance needs. Default. + */ + None = 10, // AAUDIO_PERFORMANCE_MODE_NONE, + + /** + * Extending battery life is most important. + */ + PowerSaving = 11, // AAUDIO_PERFORMANCE_MODE_POWER_SAVING, + + /** + * Reducing latency is most important. + */ + LowLatency = 12, // AAUDIO_PERFORMANCE_MODE_LOW_LATENCY + }; + + /** + * The underlying audio API used by the audio stream. + */ + enum class AudioApi : int32_t { + /** + * Try to use AAudio. If not available then use OpenSL ES. + */ + Unspecified = kUnspecified, + + /** + * Use OpenSL ES. + */ + OpenSLES, + + /** + * Try to use AAudio. Fail if unavailable. + */ + AAudio + }; + + /** + * Specifies the quality of the sample rate conversion performed by Oboe. + * Higher quality will require more CPU load. + * Higher quality conversion will probably be implemented using a sinc based resampler. + */ + enum class SampleRateConversionQuality : int32_t { + /** + * No conversion by Oboe. Underlying APIs may still do conversion. + */ + None, + /** + * Fastest conversion but may not sound great. + * This may be implemented using bilinear interpolation. + */ + Fastest, + Low, + Medium, + High, + /** + * Highest quality conversion, which may be expensive in terms of CPU. + */ + Best, + }; + + /** + * The Usage attribute expresses *why* you are playing a sound, what is this sound used for. + * This information is used by certain platforms or routing policies + * to make more refined volume or routing decisions. + * + * Note that these match the equivalent values in AudioAttributes in the Android Java API. + * + * This attribute only has an effect on Android API 28+. + */ + enum class Usage : int32_t { // aaudio_usage_t + /** + * Use this for streaming media, music performance, video, podcasts, etcetera. + */ + Media = 1, // AAUDIO_USAGE_MEDIA + + /** + * Use this for voice over IP, telephony, etcetera. + */ + VoiceCommunication = 2, // AAUDIO_USAGE_VOICE_COMMUNICATION + + /** + * Use this for sounds associated with telephony such as busy tones, DTMF, etcetera. + */ + VoiceCommunicationSignalling = 3, // AAUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING + + /** + * Use this to demand the users attention. + */ + Alarm = 4, // AAUDIO_USAGE_ALARM + + /** + * Use this for notifying the user when a message has arrived or some + * other background event has occured. + */ + Notification = 5, // AAUDIO_USAGE_NOTIFICATION + + /** + * Use this when the phone rings. + */ + NotificationRingtone = 6, // AAUDIO_USAGE_NOTIFICATION_RINGTONE + + /** + * Use this to attract the users attention when, for example, the battery is low. + */ + NotificationEvent = 10, // AAUDIO_USAGE_NOTIFICATION_EVENT + + /** + * Use this for screen readers, etcetera. + */ + AssistanceAccessibility = 11, // AAUDIO_USAGE_ASSISTANCE_ACCESSIBILITY + + /** + * Use this for driving or navigation directions. + */ + AssistanceNavigationGuidance = 12, // AAUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE + + /** + * Use this for user interface sounds, beeps, etcetera. + */ + AssistanceSonification = 13, // AAUDIO_USAGE_ASSISTANCE_SONIFICATION + + /** + * Use this for game audio and sound effects. + */ + Game = 14, // AAUDIO_USAGE_GAME + + /** + * Use this for audio responses to user queries, audio instructions or help utterances. + */ + Assistant = 16, // AAUDIO_USAGE_ASSISTANT + }; + + + /** + * The ContentType attribute describes *what* you are playing. + * It expresses the general category of the content. This information is optional. + * But in case it is known (for instance {@link Movie} for a + * movie streaming service or {@link Speech} for + * an audio book application) this information might be used by the audio framework to + * enforce audio focus. + * + * Note that these match the equivalent values in AudioAttributes in the Android Java API. + * + * This attribute only has an effect on Android API 28+. + */ + enum ContentType : int32_t { // aaudio_content_type_t + + /** + * Use this for spoken voice, audio books, etcetera. + */ + Speech = 1, // AAUDIO_CONTENT_TYPE_SPEECH + + /** + * Use this for pre-recorded or live music. + */ + Music = 2, // AAUDIO_CONTENT_TYPE_MUSIC + + /** + * Use this for a movie or video soundtrack. + */ + Movie = 3, // AAUDIO_CONTENT_TYPE_MOVIE + + /** + * Use this for sound is designed to accompany a user action, + * such as a click or beep sound made when the user presses a button. + */ + Sonification = 4, // AAUDIO_CONTENT_TYPE_SONIFICATION + }; + + /** + * Defines the audio source. + * An audio source defines both a default physical source of audio signal, and a recording + * configuration. + * + * Note that these match the equivalent values in MediaRecorder.AudioSource in the Android Java API. + * + * This attribute only has an effect on Android API 28+. + */ + enum InputPreset : int32_t { // aaudio_input_preset_t + /** + * Use this preset when other presets do not apply. + */ + Generic = 1, // AAUDIO_INPUT_PRESET_GENERIC + + /** + * Use this preset when recording video. + */ + Camcorder = 5, // AAUDIO_INPUT_PRESET_CAMCORDER + + /** + * Use this preset when doing speech recognition. + */ + VoiceRecognition = 6, // AAUDIO_INPUT_PRESET_VOICE_RECOGNITION + + /** + * Use this preset when doing telephony or voice messaging. + */ + VoiceCommunication = 7, // AAUDIO_INPUT_PRESET_VOICE_COMMUNICATION + + /** + * Use this preset to obtain an input with no effects. + * Note that this input will not have automatic gain control + * so the recorded volume may be very low. + */ + Unprocessed = 9, // AAUDIO_INPUT_PRESET_UNPROCESSED + + /** + * Use this preset for capturing audio meant to be processed in real time + * and played back for live performance (e.g karaoke). + * The capture path will minimize latency and coupling with playback path. + */ + VoicePerformance = 10, // AAUDIO_INPUT_PRESET_VOICE_PERFORMANCE + + }; + + /** + * This attribute can be used to allocate a session ID to the audio stream. + * + * This attribute only has an effect on Android API 28+. + */ + enum SessionId { + /** + * Do not allocate a session ID. + * Effects cannot be used with this stream. + * Default. + */ + None = -1, // AAUDIO_SESSION_ID_NONE + + /** + * Allocate a session ID that can be used to attach and control + * effects using the Java AudioEffects API. + * Note that the use of this flag may result in higher latency. + * + * Note that this matches the value of AudioManager.AUDIO_SESSION_ID_GENERATE. + */ + Allocate = 0, // AAUDIO_SESSION_ID_ALLOCATE + }; + + /** + * The channel count of the audio stream. The underlying type is `int32_t`. + * Use of this enum is convenient to avoid "magic" + * numbers when specifying the channel count. + * + * For example, you can write + * `builder.setChannelCount(ChannelCount::Stereo)` + * rather than `builder.setChannelCount(2)` + * + */ + enum ChannelCount : int32_t { + /** + * Audio channel count definition, use Mono or Stereo + */ + Unspecified = kUnspecified, + + /** + * Use this for mono audio + */ + Mono = 1, + + /** + * Use this for stereo audio. + */ + Stereo = 2, + }; + + /** + * On API 16 to 26 OpenSL ES will be used. When using OpenSL ES the optimal values for sampleRate and + * framesPerBurst are not known by the native code. + * On API 17+ these values should be obtained from the AudioManager using this code: + * + * 

+     * // Note that this technique only works for built-in speakers and headphones.
+     * AudioManager myAudioMgr = (AudioManager) getSystemService(Context.AUDIO_SERVICE);
+     * String sampleRateStr = myAudioMgr.getProperty(AudioManager.PROPERTY_OUTPUT_SAMPLE_RATE);
+     * int defaultSampleRate = Integer.parseInt(sampleRateStr);
+     * String framesPerBurstStr = myAudioMgr.getProperty(AudioManager.PROPERTY_OUTPUT_FRAMES_PER_BUFFER);
+     * int defaultFramesPerBurst = Integer.parseInt(framesPerBurstStr);
+     *
+ * + * It can then be passed down to Oboe through JNI. + * + * AAudio will get the optimal framesPerBurst from the HAL and will ignore this value. + */ + class DefaultStreamValues { + + public: + + /** The default sample rate to use when opening new audio streams */ + static int32_t SampleRate; + /** The default frames per burst to use when opening new audio streams */ + static int32_t FramesPerBurst; + /** The default channel count to use when opening new audio streams */ + static int32_t ChannelCount; + + }; + + /** + * The time at which the frame at `position` was presented + */ + struct FrameTimestamp { + int64_t position; // in frames + int64_t timestamp; // in nanoseconds + }; + +} // namespace oboe + +#endif // OBOE_DEFINITIONS_H diff --git a/modules/juce_audio_devices/native/oboe/include/oboe/LatencyTuner.h b/modules/juce_audio_devices/native/oboe/include/oboe/LatencyTuner.h new file mode 100644 index 0000000000..df472f56d7 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/include/oboe/LatencyTuner.h @@ -0,0 +1,118 @@ +/* + * Copyright 2017 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_LATENCY_TUNER_ +#define OBOE_LATENCY_TUNER_ + +#include +#include +#include "oboe/Definitions.h" +#include "oboe/AudioStream.h" + +namespace oboe { + +/** + * LatencyTuner can be used to dynamically tune the latency of an output stream. + * It adjusts the stream's bufferSize by monitoring the number of underruns. + * + * This only affects the latency associated with the first level of buffering that is closest + * to the application. It does not affect low latency in the HAL, or touch latency in the UI. + * + * Call tune() right before returning from your data callback function if using callbacks. + * Call tune() right before calling write() if using blocking writes. + * + * If you want to see the ongoing results of this tuning process then call + * stream->getBufferSize() periodically. + * + */ +class LatencyTuner { +public: + + /** + * Construct a new LatencyTuner object which will act on the given audio stream + * + * @param stream the stream who's latency will be tuned + */ + explicit LatencyTuner(AudioStream &stream); + + /** + * Construct a new LatencyTuner object which will act on the given audio stream. + * + * @param stream the stream who's latency will be tuned + * @param the maximum buffer size which the tune() operation will set the buffer size to + */ + explicit LatencyTuner(AudioStream &stream, int32_t maximumBufferSize); + + /** + * Adjust the bufferSizeInFrames to optimize latency. + * It will start with a low latency and then raise it if an underrun occurs. + * + * Latency tuning is only supported for AAudio. + * + * @return OK or negative error, ErrorUnimplemented for OpenSL ES + */ + Result tune(); + + /** + * This may be called from another thread. Then tune() will call reset(), + * which will lower the latency to the minimum and then allow it to rise back up + * if there are glitches. + * + * This is typically called in response to a user decision to minimize latency. In other words, + * call this from a button handler. + */ + void requestReset(); + + /** + * @return true if the audio stream's buffer size is at the maximum value. If no maximum value + * was specified when constructing the LatencyTuner then the value of + * stream->getBufferCapacityInFrames is used + */ + bool isAtMaximumBufferSize(); + + +private: + + /** + * Drop the latency down to the minimum and then let it rise back up. + * This is useful if a glitch caused the latency to increase and it hasn't gone back down. + * + * This should only be called in the same thread as tune(). + */ + void reset(); + + enum class State { + Idle, + Active, + AtMax, + Unsupported + } ; + + // arbitrary number of calls to wait before bumping up the latency + static constexpr int32_t kIdleCount = 8; + + AudioStream &mStream; + State mState = State::Idle; + int32_t mMaxBufferSize = 0; + int32_t mPreviousXRuns = 0; + int32_t mIdleCountDown = 0; + std::atomic mLatencyTriggerRequests{0}; // TODO user atomic requester from AAudio + std::atomic mLatencyTriggerResponses{0}; +}; + +} // namespace oboe + +#endif // OBOE_LATENCY_TUNER_ diff --git a/modules/juce_audio_devices/native/oboe/include/oboe/Oboe.h b/modules/juce_audio_devices/native/oboe/include/oboe/Oboe.h new file mode 100644 index 0000000000..c3a0bcabdf --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/include/oboe/Oboe.h @@ -0,0 +1,37 @@ +/* + * Copyright (C) 2016 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_OBOE_H +#define OBOE_OBOE_H + +/** + * \mainpage API reference + * + * All documentation is found in the oboe namespace section + * + */ + +#include "oboe/Definitions.h" +#include "oboe/ResultWithValue.h" +#include "oboe/LatencyTuner.h" +#include "oboe/AudioStream.h" +#include "oboe/AudioStreamBase.h" +#include "oboe/AudioStreamBuilder.h" +#include "oboe/Utilities.h" +#include "oboe/Version.h" +#include "oboe/StabilizedCallback.h" + +#endif //OBOE_OBOE_H diff --git a/modules/juce_audio_devices/native/oboe/include/oboe/ResultWithValue.h b/modules/juce_audio_devices/native/oboe/include/oboe/ResultWithValue.h new file mode 100644 index 0000000000..fcb1fac141 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/include/oboe/ResultWithValue.h @@ -0,0 +1,155 @@ +/* + * Copyright (C) 2018 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_RESULT_WITH_VALUE_H +#define OBOE_RESULT_WITH_VALUE_H + +#include "oboe/Definitions.h" +#include +#include + +namespace oboe { + +/** + * A ResultWithValue can store both the result of an operation (either OK or an error) and a value. + * + * It has been designed for cases where the caller needs to know whether an operation succeeded and, + * if it did, a value which was obtained during the operation. + * + * For example, when reading from a stream the caller needs to know the result of the read operation + * and, if it was successful, how many frames were read. Note that ResultWithValue can be evaluated + * as a boolean so it's simple to check whether the result is OK. + * + * + * ResultWithValue resultOfRead = myStream.read(&buffer, numFrames, timeoutNanoseconds); + * + * if (resultOfRead) { + * LOGD("Frames read: %d", resultOfRead.value()); + * } else { + * LOGD("Error reading from stream: %s", resultOfRead.error()); + * } + * + */ +template +class ResultWithValue { +public: + + /** + * Construct a ResultWithValue containing an error result. + * + * @param error The error + */ + ResultWithValue(oboe::Result error) + : mValue{} + , mError(error) {} + + /** + * Construct a ResultWithValue containing an OK result and a value. + * + * @param value the value to store + */ + explicit ResultWithValue(T value) + : mValue(value) + , mError(oboe::Result::OK) {} + + /** + * Get the result. + * + * @return the result + */ + oboe::Result error() const { + return mError; + } + + /** + * Get the value + * @return + */ + T value() const { + return mValue; + } + + /** + * @return true if OK + */ + explicit operator bool() const { return mError == oboe::Result::OK; } + + /** + * Quick way to check for an error. + * + * The caller could write something like this: + * + * if (!result) { printf("Got error %s\n", convertToText(result.error())); } + * + * + * @return true if an error occurred + */ + bool operator !() const { return mError != oboe::Result::OK; } + + /** + * Implicitly convert to a Result. This enables easy comparison with Result values. Example: + * + * + * ResultWithValue result = openStream(); + * if (result == Result::ErrorNoMemory){ // tell user they're out of memory } + * + */ + operator Result() const { + return mError; + } + + /** + * Create a ResultWithValue from a number. If the number is positive the ResultWithValue will + * have a result of Result::OK and the value will contain the number. If the number is negative + * the result will be obtained from the negative number (numeric error codes can be found in + * AAudio.h) and the value will be null. + * + */ + static ResultWithValue createBasedOnSign(T numericResult){ + + // Ensure that the type is either an integer or float + static_assert(std::is_arithmetic::value, + "createBasedOnSign can only be called for numeric types (int or float)"); + + if (numericResult >= 0){ + return ResultWithValue(numericResult); + } else { + return ResultWithValue(static_cast(numericResult)); + } + } + +private: + const T mValue; + const oboe::Result mError; +}; + +/** + * If the result is `OK` then return the value, otherwise return a human-readable error message. + */ +template +std::ostream& operator<<(std::ostream &strm, const ResultWithValue &result) { + if (!result) { + strm << convertToText(result.error()); + } else { + strm << result.value(); + } + return strm; +} + +} // namespace oboe + + +#endif //OBOE_RESULT_WITH_VALUE_H diff --git a/modules/juce_audio_devices/native/oboe/include/oboe/StabilizedCallback.h b/modules/juce_audio_devices/native/oboe/include/oboe/StabilizedCallback.h new file mode 100644 index 0000000000..3f1a689690 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/include/oboe/StabilizedCallback.h @@ -0,0 +1,75 @@ +/* + * Copyright 2018 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_STABILIZEDCALLBACK_H +#define OBOE_STABILIZEDCALLBACK_H + +#include +#include "oboe/AudioStream.h" + +namespace oboe { + +class StabilizedCallback : public AudioStreamCallback { + +public: + explicit StabilizedCallback(AudioStreamCallback *callback); + + DataCallbackResult + onAudioReady(AudioStream *oboeStream, void *audioData, int32_t numFrames) override; + + void onErrorBeforeClose(AudioStream *oboeStream, Result error) override { + return mCallback->onErrorBeforeClose(oboeStream, error); + } + + void onErrorAfterClose(AudioStream *oboeStream, Result error) override { + + // Reset all fields now that the stream has been closed + mFrameCount = 0; + mEpochTimeNanos = 0; + mOpsPerNano = 1; + return mCallback->onErrorAfterClose(oboeStream, error); + } + +private: + + AudioStreamCallback *mCallback = nullptr; + int64_t mFrameCount = 0; + int64_t mEpochTimeNanos = 0; + double mOpsPerNano = 1; + + void generateLoad(int64_t durationNanos); +}; + +/** + * cpu_relax is an architecture specific method of telling the CPU that you don't want it to + * do much work. asm volatile keeps the compiler from optimising these instructions out. + */ +#if defined(__i386__) || defined(__x86_64__) +#define cpu_relax() asm volatile("rep; nop" ::: "memory"); + +#elif defined(__arm__) || defined(__mips__) + #define cpu_relax() asm volatile("":::"memory") + +#elif defined(__aarch64__) +#define cpu_relax() asm volatile("yield" ::: "memory") + +#else +#error "cpu_relax is not defined for this architecture" +#endif + +} + +#endif //OBOE_STABILIZEDCALLBACK_H diff --git a/modules/juce_audio_devices/native/oboe/include/oboe/Utilities.h b/modules/juce_audio_devices/native/oboe/include/oboe/Utilities.h new file mode 100644 index 0000000000..72389917ce --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/include/oboe/Utilities.h @@ -0,0 +1,73 @@ +/* + * Copyright 2016 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_UTILITIES_H +#define OBOE_UTILITIES_H + +#include +#include +#include "oboe/Definitions.h" + +namespace oboe { + +/** + * Convert an array of floats to an array of 16-bit integers. + * + * @param source the input array. + * @param destination the output array. + * @param numSamples the number of values to convert. + */ +void convertFloatToPcm16(const float *source, int16_t *destination, int32_t numSamples); + +/** + * Convert an array of 16-bit integers to an array of floats. + * + * @param source the input array. + * @param destination the output array. + * @param numSamples the number of values to convert. + */ +void convertPcm16ToFloat(const int16_t *source, float *destination, int32_t numSamples); + +/** + * @return the size of a sample of the given format in bytes or 0 if format is invalid + */ +int32_t convertFormatToSizeInBytes(AudioFormat format); + +/** + * The text is the ASCII symbol corresponding to the supplied Oboe enum value, + * or an English message saying the value is unrecognized. + * This is intended for developers to use when debugging. + * It is not for displaying to users. + * + * @param input object to convert from. @see common/Utilities.cpp for concrete implementations + * @return text representation of an Oboe enum value. There is no need to call free on this. + */ +template +const char * convertToText(FromType input); + +/** + * Return the version of the SDK that is currently running. + * + * For example, on Android, this would return 27 for Oreo 8.1. + * If the version number cannot be determined then this will return -1. + * + * @return version number or -1 + */ +int getSdkVersion(); + +} // namespace oboe + +#endif //OBOE_UTILITIES_H diff --git a/modules/juce_audio_devices/native/oboe/include/oboe/Version.h b/modules/juce_audio_devices/native/oboe/include/oboe/Version.h new file mode 100644 index 0000000000..24c836da6c --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/include/oboe/Version.h @@ -0,0 +1,92 @@ +/* + * Copyright 2017 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_VERSIONINFO_H +#define OBOE_VERSIONINFO_H + +#include + +/** + * A note on use of preprocessor defines: + * + * This is one of the few times when it's suitable to use preprocessor defines rather than constexpr + * Why? Because C++11 requires a lot of boilerplate code to convert integers into compile-time + * string literals. The preprocessor, despite it's lack of type checking, is more suited to the task + * + * See: https://stackoverflow.com/questions/6713420/c-convert-integer-to-string-at-compile-time/26824971#26824971 + * + */ + +// Type: 8-bit unsigned int. Min value: 0 Max value: 255. See below for description. +#define OBOE_VERSION_MAJOR 1 + +// Type: 8-bit unsigned int. Min value: 0 Max value: 255. See below for description. +#define OBOE_VERSION_MINOR 2 + +// Type: 16-bit unsigned int. Min value: 0 Max value: 65535. See below for description. +#define OBOE_VERSION_PATCH 4 + +#define OBOE_STRINGIFY(x) #x +#define OBOE_TOSTRING(x) OBOE_STRINGIFY(x) + +// Type: String literal. See below for description. +#define OBOE_VERSION_TEXT \ + OBOE_TOSTRING(OBOE_VERSION_MAJOR) "." \ + OBOE_TOSTRING(OBOE_VERSION_MINOR) "." \ + OBOE_TOSTRING(OBOE_VERSION_PATCH) + +// Type: 32-bit unsigned int. See below for description. +#define OBOE_VERSION_NUMBER ((OBOE_VERSION_MAJOR << 24) | (OBOE_VERSION_MINOR << 16) | OBOE_VERSION_PATCH) + +namespace oboe { + +const char * getVersionText(); + +/** + * Oboe versioning object + */ +struct Version { + /** + * This is incremented when we make breaking API changes. Based loosely on https://semver.org/. + */ + static constexpr uint8_t Major = OBOE_VERSION_MAJOR; + + /** + * This is incremented when we add backwards compatible functionality. Or set to zero when MAJOR is + * incremented. + */ + static constexpr uint8_t Minor = OBOE_VERSION_MINOR; + + /** + * This is incremented when we make backwards compatible bug fixes. Or set to zero when MINOR is + * incremented. + */ + static constexpr uint16_t Patch = OBOE_VERSION_PATCH; + + /** + * Version string in the form MAJOR.MINOR.PATCH. + */ + static constexpr const char * Text = OBOE_VERSION_TEXT; + + /** + * Integer representation of the current Oboe library version. This will always increase when the + * version number changes so can be compared using integer comparison. + */ + static constexpr uint32_t Number = OBOE_VERSION_NUMBER; +}; + +} // namespace oboe +#endif //OBOE_VERSIONINFO_H diff --git a/modules/juce_audio_devices/native/oboe/readme.md b/modules/juce_audio_devices/native/oboe/readme.md new file mode 100644 index 0000000000..ac8c98e50d --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/readme.md @@ -0,0 +1,10 @@ +The files in this directory are reproduced from the official Oboe repository, which can be found at +github.com/google/oboe. + +These files are from tag 1.3 (5c42747). + +We've included only those parts of the original repository which are required to build the Oboe +library. Documentation, samples, tests, and other non-library items have been omitted. + +Files in this directory and below are licensed under the terms of the license in the LICENSE file +which you can find in the same directory as this readme. diff --git a/modules/juce_audio_devices/native/oboe/src/aaudio/AAudioLoader.cpp b/modules/juce_audio_devices/native/oboe/src/aaudio/AAudioLoader.cpp new file mode 100644 index 0000000000..064633c6e5 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/aaudio/AAudioLoader.cpp @@ -0,0 +1,348 @@ +/* + * Copyright 2016 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include +#include +#include "common/OboeDebug.h" +#include "AAudioLoader.h" + +#define LIB_AAUDIO_NAME "libaaudio.so" + +namespace oboe { + +AAudioLoader::~AAudioLoader() { + if (mLibHandle != nullptr) { + dlclose(mLibHandle); + mLibHandle = nullptr; + } +} + +AAudioLoader* AAudioLoader::getInstance() { + static AAudioLoader instance; + return &instance; +} + +int AAudioLoader::open() { + if (mLibHandle != nullptr) { + return 0; + } + + // Use RTLD_NOW to avoid the unpredictable behavior that RTLD_LAZY can cause. + // Also resolving all the links now will prevent a run-time penalty later. + mLibHandle = dlopen(LIB_AAUDIO_NAME, RTLD_NOW); + if (mLibHandle == nullptr) { + LOGI("AAudioLoader::open() could not find " LIB_AAUDIO_NAME); + return -1; // TODO review return code + } else { + LOGD("AAudioLoader(): dlopen(%s) returned %p", LIB_AAUDIO_NAME, mLibHandle); + } + + // Load all the function pointers. + createStreamBuilder = load_I_PPB("AAudio_createStreamBuilder"); + builder_openStream = load_I_PBPPS("AAudioStreamBuilder_openStream"); + + builder_setChannelCount = load_V_PBI("AAudioStreamBuilder_setChannelCount"); + if (builder_setChannelCount == nullptr) { + // Use old deprecated alias if needed. + builder_setChannelCount = load_V_PBI("AAudioStreamBuilder_setSamplesPerFrame"); + } + + builder_setBufferCapacityInFrames = load_V_PBI("AAudioStreamBuilder_setBufferCapacityInFrames"); + builder_setDeviceId = load_V_PBI("AAudioStreamBuilder_setDeviceId"); + builder_setDirection = load_V_PBI("AAudioStreamBuilder_setDirection"); + builder_setFormat = load_V_PBI("AAudioStreamBuilder_setFormat"); + builder_setFramesPerDataCallback = load_V_PBI("AAudioStreamBuilder_setFramesPerDataCallback"); + builder_setSharingMode = load_V_PBI("AAudioStreamBuilder_setSharingMode"); + builder_setPerformanceMode = load_V_PBI("AAudioStreamBuilder_setPerformanceMode"); + builder_setSampleRate = load_V_PBI("AAudioStreamBuilder_setSampleRate"); + + if (getSdkVersion() >= __ANDROID_API_P__){ + builder_setUsage = load_V_PBI("AAudioStreamBuilder_setUsage"); + builder_setContentType = load_V_PBI("AAudioStreamBuilder_setContentType"); + builder_setInputPreset = load_V_PBI("AAudioStreamBuilder_setInputPreset"); + builder_setSessionId = load_V_PBI("AAudioStreamBuilder_setSessionId"); + } + + builder_delete = load_I_PB("AAudioStreamBuilder_delete"); + + + builder_setDataCallback = load_V_PBPDPV("AAudioStreamBuilder_setDataCallback"); + builder_setErrorCallback = load_V_PBPEPV("AAudioStreamBuilder_setErrorCallback"); + + stream_read = load_I_PSPVIL("AAudioStream_read"); + + stream_write = load_I_PSCPVIL("AAudioStream_write"); + + stream_waitForStateChange = load_I_PSTPTL("AAudioStream_waitForStateChange"); + + stream_getTimestamp = load_I_PSKPLPL("AAudioStream_getTimestamp"); + + stream_isMMapUsed = load_B_PS("AAudioStream_isMMapUsed"); + + stream_getChannelCount = load_I_PS("AAudioStream_getChannelCount"); + if (stream_getChannelCount == nullptr) { + // Use old alias if needed. + stream_getChannelCount = load_I_PS("AAudioStream_getSamplesPerFrame"); + } + + stream_close = load_I_PS("AAudioStream_close"); + + stream_getBufferSize = load_I_PS("AAudioStream_getBufferSizeInFrames"); + stream_getDeviceId = load_I_PS("AAudioStream_getDeviceId"); + stream_getBufferCapacity = load_I_PS("AAudioStream_getBufferCapacityInFrames"); + stream_getFormat = load_F_PS("AAudioStream_getFormat"); + stream_getFramesPerBurst = load_I_PS("AAudioStream_getFramesPerBurst"); + stream_getFramesRead = load_L_PS("AAudioStream_getFramesRead"); + stream_getFramesWritten = load_L_PS("AAudioStream_getFramesWritten"); + stream_getPerformanceMode = load_I_PS("AAudioStream_getPerformanceMode"); + stream_getSampleRate = load_I_PS("AAudioStream_getSampleRate"); + stream_getSharingMode = load_I_PS("AAudioStream_getSharingMode"); + stream_getState = load_I_PS("AAudioStream_getState"); + stream_getXRunCount = load_I_PS("AAudioStream_getXRunCount"); + + stream_requestStart = load_I_PS("AAudioStream_requestStart"); + stream_requestPause = load_I_PS("AAudioStream_requestPause"); + stream_requestFlush = load_I_PS("AAudioStream_requestFlush"); + stream_requestStop = load_I_PS("AAudioStream_requestStop"); + + stream_setBufferSize = load_I_PSI("AAudioStream_setBufferSizeInFrames"); + + convertResultToText = load_CPH_I("AAudio_convertResultToText"); + + if (getSdkVersion() >= __ANDROID_API_P__){ + stream_getUsage = load_I_PS("AAudioStream_getUsage"); + stream_getContentType = load_I_PS("AAudioStream_getContentType"); + stream_getInputPreset = load_I_PS("AAudioStream_getInputPreset"); + stream_getSessionId = load_I_PS("AAudioStream_getSessionId"); + } + return 0; +} + +static void AAudioLoader_check(void *proc, const char *functionName) { + if (proc == nullptr) { + LOGW("AAudioLoader could not find %s", functionName); + } +} + +AAudioLoader::signature_I_PPB AAudioLoader::load_I_PPB(const char *functionName) { + void *proc = dlsym(mLibHandle, functionName); + AAudioLoader_check(proc, functionName); + return reinterpret_cast(proc); +} + +AAudioLoader::signature_CPH_I AAudioLoader::load_CPH_I(const char *functionName) { + void *proc = dlsym(mLibHandle, functionName); + AAudioLoader_check(proc, functionName); + return reinterpret_cast(proc); +} + +AAudioLoader::signature_V_PBI AAudioLoader::load_V_PBI(const char *functionName) { + void *proc = dlsym(mLibHandle, functionName); + AAudioLoader_check(proc, functionName); + return reinterpret_cast(proc); +} + +AAudioLoader::signature_V_PBPDPV AAudioLoader::load_V_PBPDPV(const char *functionName) { + void *proc = dlsym(mLibHandle, functionName); + AAudioLoader_check(proc, functionName); + return reinterpret_cast(proc); +} + +AAudioLoader::signature_V_PBPEPV AAudioLoader::load_V_PBPEPV(const char *functionName) { + void *proc = dlsym(mLibHandle, functionName); + AAudioLoader_check(proc, functionName); + return reinterpret_cast(proc); +} + +AAudioLoader::signature_I_PSI AAudioLoader::load_I_PSI(const char *functionName) { + void *proc = dlsym(mLibHandle, functionName); + AAudioLoader_check(proc, functionName); + return reinterpret_cast(proc); +} + +AAudioLoader::signature_I_PS AAudioLoader::load_I_PS(const char *functionName) { + void *proc = dlsym(mLibHandle, functionName); + AAudioLoader_check(proc, functionName); + return reinterpret_cast(proc); +} + +AAudioLoader::signature_L_PS AAudioLoader::load_L_PS(const char *functionName) { + void *proc = dlsym(mLibHandle, functionName); + AAudioLoader_check(proc, functionName); + return reinterpret_cast(proc); +} + +AAudioLoader::signature_F_PS AAudioLoader::load_F_PS(const char *functionName) { + void *proc = dlsym(mLibHandle, functionName); + AAudioLoader_check(proc, functionName); + return reinterpret_cast(proc); +} + +AAudioLoader::signature_B_PS AAudioLoader::load_B_PS(const char *functionName) { + void *proc = dlsym(mLibHandle, functionName); + AAudioLoader_check(proc, functionName); + return reinterpret_cast(proc); +} + +AAudioLoader::signature_I_PB AAudioLoader::load_I_PB(const char *functionName) { + void *proc = dlsym(mLibHandle, functionName); + AAudioLoader_check(proc, functionName); + return reinterpret_cast(proc); +} + +AAudioLoader::signature_I_PBPPS AAudioLoader::load_I_PBPPS(const char *functionName) { + void *proc = dlsym(mLibHandle, functionName); + AAudioLoader_check(proc, functionName); + return reinterpret_cast(proc); +} + +AAudioLoader::signature_I_PSCPVIL AAudioLoader::load_I_PSCPVIL(const char *functionName) { + void *proc = dlsym(mLibHandle, functionName); + AAudioLoader_check(proc, functionName); + return reinterpret_cast(proc); +} + +AAudioLoader::signature_I_PSPVIL AAudioLoader::load_I_PSPVIL(const char *functionName) { + void *proc = dlsym(mLibHandle, functionName); + AAudioLoader_check(proc, functionName); + return reinterpret_cast(proc); +} + +AAudioLoader::signature_I_PSTPTL AAudioLoader::load_I_PSTPTL(const char *functionName) { + void *proc = dlsym(mLibHandle, functionName); + AAudioLoader_check(proc, functionName); + return reinterpret_cast(proc); +} + +AAudioLoader::signature_I_PSKPLPL AAudioLoader::load_I_PSKPLPL(const char *functionName) { + void *proc = dlsym(mLibHandle, functionName); + AAudioLoader_check(proc, functionName); + return reinterpret_cast(proc); +} + +// Ensure that all AAudio primitive data types are int32_t +#define ASSERT_INT32(type) static_assert(std::is_same::value, \ +#type" must be int32_t") + +#define ERRMSG "Oboe constants must match AAudio constants." + +// These asserts help verify that the Oboe definitions match the equivalent AAudio definitions. +// This code is in this .cpp file so it only gets tested once. +#ifdef AAUDIO_AAUDIO_H + + ASSERT_INT32(aaudio_stream_state_t); + ASSERT_INT32(aaudio_direction_t); + ASSERT_INT32(aaudio_format_t); + ASSERT_INT32(aaudio_data_callback_result_t); + ASSERT_INT32(aaudio_result_t); + ASSERT_INT32(aaudio_sharing_mode_t); + ASSERT_INT32(aaudio_performance_mode_t); + + static_assert((int32_t)StreamState::Uninitialized == AAUDIO_STREAM_STATE_UNINITIALIZED, ERRMSG); + static_assert((int32_t)StreamState::Unknown == AAUDIO_STREAM_STATE_UNKNOWN, ERRMSG); + static_assert((int32_t)StreamState::Open == AAUDIO_STREAM_STATE_OPEN, ERRMSG); + static_assert((int32_t)StreamState::Starting == AAUDIO_STREAM_STATE_STARTING, ERRMSG); + static_assert((int32_t)StreamState::Started == AAUDIO_STREAM_STATE_STARTED, ERRMSG); + static_assert((int32_t)StreamState::Pausing == AAUDIO_STREAM_STATE_PAUSING, ERRMSG); + static_assert((int32_t)StreamState::Paused == AAUDIO_STREAM_STATE_PAUSED, ERRMSG); + static_assert((int32_t)StreamState::Flushing == AAUDIO_STREAM_STATE_FLUSHING, ERRMSG); + static_assert((int32_t)StreamState::Flushed == AAUDIO_STREAM_STATE_FLUSHED, ERRMSG); + static_assert((int32_t)StreamState::Stopping == AAUDIO_STREAM_STATE_STOPPING, ERRMSG); + static_assert((int32_t)StreamState::Stopped == AAUDIO_STREAM_STATE_STOPPED, ERRMSG); + static_assert((int32_t)StreamState::Closing == AAUDIO_STREAM_STATE_CLOSING, ERRMSG); + static_assert((int32_t)StreamState::Closed == AAUDIO_STREAM_STATE_CLOSED, ERRMSG); + static_assert((int32_t)StreamState::Disconnected == AAUDIO_STREAM_STATE_DISCONNECTED, ERRMSG); + + static_assert((int32_t)Direction::Output == AAUDIO_DIRECTION_OUTPUT, ERRMSG); + static_assert((int32_t)Direction::Input == AAUDIO_DIRECTION_INPUT, ERRMSG); + + static_assert((int32_t)AudioFormat::Invalid == AAUDIO_FORMAT_INVALID, ERRMSG); + static_assert((int32_t)AudioFormat::Unspecified == AAUDIO_FORMAT_UNSPECIFIED, ERRMSG); + static_assert((int32_t)AudioFormat::I16 == AAUDIO_FORMAT_PCM_I16, ERRMSG); + static_assert((int32_t)AudioFormat::Float == AAUDIO_FORMAT_PCM_FLOAT, ERRMSG); + + static_assert((int32_t)DataCallbackResult::Continue == AAUDIO_CALLBACK_RESULT_CONTINUE, ERRMSG); + static_assert((int32_t)DataCallbackResult::Stop == AAUDIO_CALLBACK_RESULT_STOP, ERRMSG); + + static_assert((int32_t)Result::OK == AAUDIO_OK, ERRMSG); + static_assert((int32_t)Result::ErrorBase == AAUDIO_ERROR_BASE, ERRMSG); + static_assert((int32_t)Result::ErrorDisconnected == AAUDIO_ERROR_DISCONNECTED, ERRMSG); + static_assert((int32_t)Result::ErrorIllegalArgument == AAUDIO_ERROR_ILLEGAL_ARGUMENT, ERRMSG); + static_assert((int32_t)Result::ErrorInternal == AAUDIO_ERROR_INTERNAL, ERRMSG); + static_assert((int32_t)Result::ErrorInvalidState == AAUDIO_ERROR_INVALID_STATE, ERRMSG); + static_assert((int32_t)Result::ErrorInvalidHandle == AAUDIO_ERROR_INVALID_HANDLE, ERRMSG); + static_assert((int32_t)Result::ErrorUnimplemented == AAUDIO_ERROR_UNIMPLEMENTED, ERRMSG); + static_assert((int32_t)Result::ErrorUnavailable == AAUDIO_ERROR_UNAVAILABLE, ERRMSG); + static_assert((int32_t)Result::ErrorNoFreeHandles == AAUDIO_ERROR_NO_FREE_HANDLES, ERRMSG); + static_assert((int32_t)Result::ErrorNoMemory == AAUDIO_ERROR_NO_MEMORY, ERRMSG); + static_assert((int32_t)Result::ErrorNull == AAUDIO_ERROR_NULL, ERRMSG); + static_assert((int32_t)Result::ErrorTimeout == AAUDIO_ERROR_TIMEOUT, ERRMSG); + static_assert((int32_t)Result::ErrorWouldBlock == AAUDIO_ERROR_WOULD_BLOCK, ERRMSG); + static_assert((int32_t)Result::ErrorInvalidFormat == AAUDIO_ERROR_INVALID_FORMAT, ERRMSG); + static_assert((int32_t)Result::ErrorOutOfRange == AAUDIO_ERROR_OUT_OF_RANGE, ERRMSG); + static_assert((int32_t)Result::ErrorNoService == AAUDIO_ERROR_NO_SERVICE, ERRMSG); + static_assert((int32_t)Result::ErrorInvalidRate == AAUDIO_ERROR_INVALID_RATE, ERRMSG); + + static_assert((int32_t)SharingMode::Exclusive == AAUDIO_SHARING_MODE_EXCLUSIVE, ERRMSG); + static_assert((int32_t)SharingMode::Shared == AAUDIO_SHARING_MODE_SHARED, ERRMSG); + + static_assert((int32_t)PerformanceMode::None == AAUDIO_PERFORMANCE_MODE_NONE, ERRMSG); + static_assert((int32_t)PerformanceMode::PowerSaving + == AAUDIO_PERFORMANCE_MODE_POWER_SAVING, ERRMSG); + static_assert((int32_t)PerformanceMode::LowLatency + == AAUDIO_PERFORMANCE_MODE_LOW_LATENCY, ERRMSG); +#endif + +// The aaudio_ usage, content and input_preset types were added in NDK 17, +// which is the first version to support Android Pie (API 28). +#if __NDK_MAJOR__ >= 17 + + ASSERT_INT32(aaudio_usage_t); + ASSERT_INT32(aaudio_content_type_t); + ASSERT_INT32(aaudio_input_preset_t); + + static_assert((int32_t)Usage::Media == AAUDIO_USAGE_MEDIA, ERRMSG); + static_assert((int32_t)Usage::VoiceCommunication == AAUDIO_USAGE_VOICE_COMMUNICATION, ERRMSG); + static_assert((int32_t)Usage::VoiceCommunicationSignalling + == AAUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING, ERRMSG); + static_assert((int32_t)Usage::Alarm == AAUDIO_USAGE_ALARM, ERRMSG); + static_assert((int32_t)Usage::Notification == AAUDIO_USAGE_NOTIFICATION, ERRMSG); + static_assert((int32_t)Usage::NotificationRingtone == AAUDIO_USAGE_NOTIFICATION_RINGTONE, ERRMSG); + static_assert((int32_t)Usage::NotificationEvent == AAUDIO_USAGE_NOTIFICATION_EVENT, ERRMSG); + static_assert((int32_t)Usage::AssistanceAccessibility == AAUDIO_USAGE_ASSISTANCE_ACCESSIBILITY, ERRMSG); + static_assert((int32_t)Usage::AssistanceNavigationGuidance + == AAUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE, ERRMSG); + static_assert((int32_t)Usage::AssistanceSonification == AAUDIO_USAGE_ASSISTANCE_SONIFICATION, ERRMSG); + static_assert((int32_t)Usage::Game == AAUDIO_USAGE_GAME, ERRMSG); + static_assert((int32_t)Usage::Assistant == AAUDIO_USAGE_ASSISTANT, ERRMSG); + + static_assert((int32_t)ContentType::Speech == AAUDIO_CONTENT_TYPE_SPEECH, ERRMSG); + static_assert((int32_t)ContentType::Music == AAUDIO_CONTENT_TYPE_MUSIC, ERRMSG); + static_assert((int32_t)ContentType::Movie == AAUDIO_CONTENT_TYPE_MOVIE, ERRMSG); + static_assert((int32_t)ContentType::Sonification == AAUDIO_CONTENT_TYPE_SONIFICATION, ERRMSG); + + static_assert((int32_t)InputPreset::Generic == AAUDIO_INPUT_PRESET_GENERIC, ERRMSG); + static_assert((int32_t)InputPreset::Camcorder == AAUDIO_INPUT_PRESET_CAMCORDER, ERRMSG); + static_assert((int32_t)InputPreset::VoiceRecognition == AAUDIO_INPUT_PRESET_VOICE_RECOGNITION, ERRMSG); + static_assert((int32_t)InputPreset::VoiceCommunication + == AAUDIO_INPUT_PRESET_VOICE_COMMUNICATION, ERRMSG); + static_assert((int32_t)InputPreset::Unprocessed == AAUDIO_INPUT_PRESET_UNPROCESSED, ERRMSG); + + static_assert((int32_t)SessionId::None == AAUDIO_SESSION_ID_NONE, ERRMSG); + static_assert((int32_t)SessionId::Allocate == AAUDIO_SESSION_ID_ALLOCATE, ERRMSG); +#endif + +} // namespace oboe diff --git a/modules/juce_audio_devices/native/oboe/src/aaudio/AAudioLoader.h b/modules/juce_audio_devices/native/oboe/src/aaudio/AAudioLoader.h new file mode 100644 index 0000000000..6f5bb9502f --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/aaudio/AAudioLoader.h @@ -0,0 +1,229 @@ +/* + * Copyright 2016 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_AAUDIO_LOADER_H_ +#define OBOE_AAUDIO_LOADER_H_ + +#include +#include "oboe/Definitions.h" + +// If the NDK is before O then define this in your build +// so that AAudio.h will not be included. +#ifdef OBOE_NO_INCLUDE_AAUDIO + +// Define missing types from AAudio.h +typedef int32_t aaudio_stream_state_t; +typedef int32_t aaudio_direction_t; +typedef int32_t aaudio_format_t; +typedef int32_t aaudio_data_callback_result_t; +typedef int32_t aaudio_result_t; +typedef int32_t aaudio_sharing_mode_t; +typedef int32_t aaudio_performance_mode_t; + +typedef struct AAudioStreamStruct AAudioStream; +typedef struct AAudioStreamBuilderStruct AAudioStreamBuilder; + +typedef aaudio_data_callback_result_t (*AAudioStream_dataCallback)( + AAudioStream *stream, + void *userData, + void *audioData, + int32_t numFrames); + +typedef void (*AAudioStream_errorCallback)( + AAudioStream *stream, + void *userData, + aaudio_result_t error); + +// These were defined in P +typedef int32_t aaudio_usage_t; +typedef int32_t aaudio_content_type_t; +typedef int32_t aaudio_input_preset_t; +typedef int32_t aaudio_session_id_t; +#else +#include +#include +#endif + +#ifndef __NDK_MAJOR__ +#define __NDK_MAJOR__ 0 +#endif + +namespace oboe { + + +/** + * The AAudio API was not available in early versions of Android. + * To avoid linker errors, we dynamically link with the functions by name using dlsym(). + * On older versions this linkage will safely fail. + */ +class AAudioLoader { + public: + // Use signatures for common functions. + // Key to letter abbreviations. + // S = Stream + // B = Builder + // I = int32_t + // L = int64_t + // T = sTate + // K = clocKid_t + // P = Pointer to following data type + // C = Const prefix + // H = cHar + typedef int32_t (*signature_I_PPB)(AAudioStreamBuilder **builder); + + typedef const char * (*signature_CPH_I)(int32_t); + + typedef int32_t (*signature_I_PBPPS)(AAudioStreamBuilder *, + AAudioStream **stream); // AAudioStreamBuilder_open() + + typedef int32_t (*signature_I_PB)(AAudioStreamBuilder *); // AAudioStreamBuilder_delete() + // AAudioStreamBuilder_setSampleRate() + typedef void (*signature_V_PBI)(AAudioStreamBuilder *, int32_t); + + typedef int32_t (*signature_I_PS)(AAudioStream *); // AAudioStream_getSampleRate() + typedef int64_t (*signature_L_PS)(AAudioStream *); // AAudioStream_getFramesRead() + // AAudioStream_setBufferSizeInFrames() + typedef int32_t (*signature_I_PSI)(AAudioStream *, int32_t); + + typedef void (*signature_V_PBPDPV)(AAudioStreamBuilder *, + AAudioStream_dataCallback, + void *); + + typedef void (*signature_V_PBPEPV)(AAudioStreamBuilder *, + AAudioStream_errorCallback, + void *); + + typedef aaudio_format_t (*signature_F_PS)(AAudioStream *stream); + + typedef int32_t (*signature_I_PSPVIL)(AAudioStream *, void *, int32_t, int64_t); + typedef int32_t (*signature_I_PSCPVIL)(AAudioStream *, const void *, int32_t, int64_t); + + typedef int32_t (*signature_I_PSTPTL)(AAudioStream *, + aaudio_stream_state_t, + aaudio_stream_state_t *, + int64_t); + + typedef int32_t (*signature_I_PSKPLPL)(AAudioStream *, clockid_t, int64_t *, int64_t *); + + typedef bool (*signature_B_PS)(AAudioStream *); + + static AAudioLoader* getInstance(); // singleton + + /** + * Open the AAudio shared library and load the function pointers. + * This can be called multiple times. + * It should only be called from one thread. + * + * The destructor will clean up after the open. + * + * @return 0 if successful or negative error. + */ + int open(); + + // Function pointers into the AAudio shared library. + signature_I_PPB createStreamBuilder = nullptr; + + signature_I_PBPPS builder_openStream = nullptr; + + signature_V_PBI builder_setBufferCapacityInFrames = nullptr; + signature_V_PBI builder_setChannelCount = nullptr; + signature_V_PBI builder_setDeviceId = nullptr; + signature_V_PBI builder_setDirection = nullptr; + signature_V_PBI builder_setFormat = nullptr; + signature_V_PBI builder_setFramesPerDataCallback = nullptr; + signature_V_PBI builder_setPerformanceMode = nullptr; + signature_V_PBI builder_setSampleRate = nullptr; + signature_V_PBI builder_setSharingMode = nullptr; + + signature_V_PBI builder_setUsage = nullptr; + signature_V_PBI builder_setContentType = nullptr; + signature_V_PBI builder_setInputPreset = nullptr; + signature_V_PBI builder_setSessionId = nullptr; + + signature_V_PBPDPV builder_setDataCallback = nullptr; + signature_V_PBPEPV builder_setErrorCallback = nullptr; + + signature_I_PB builder_delete = nullptr; + + signature_F_PS stream_getFormat = nullptr; + + signature_I_PSPVIL stream_read = nullptr; + signature_I_PSCPVIL stream_write = nullptr; + + signature_I_PSTPTL stream_waitForStateChange = nullptr; + + signature_I_PSKPLPL stream_getTimestamp = nullptr; + + signature_B_PS stream_isMMapUsed = nullptr; + + signature_I_PS stream_close = nullptr; + + signature_I_PS stream_getChannelCount = nullptr; + signature_I_PS stream_getDeviceId = nullptr; + + signature_I_PS stream_getBufferSize = nullptr; + signature_I_PS stream_getBufferCapacity = nullptr; + signature_I_PS stream_getFramesPerBurst = nullptr; + signature_I_PS stream_getState = nullptr; + signature_I_PS stream_getPerformanceMode = nullptr; + signature_I_PS stream_getSampleRate = nullptr; + signature_I_PS stream_getSharingMode = nullptr; + signature_I_PS stream_getXRunCount = nullptr; + + signature_I_PSI stream_setBufferSize = nullptr; + signature_I_PS stream_requestStart = nullptr; + signature_I_PS stream_requestPause = nullptr; + signature_I_PS stream_requestFlush = nullptr; + signature_I_PS stream_requestStop = nullptr; + + signature_L_PS stream_getFramesRead = nullptr; + signature_L_PS stream_getFramesWritten = nullptr; + + signature_CPH_I convertResultToText = nullptr; + + signature_I_PS stream_getUsage = nullptr; + signature_I_PS stream_getContentType = nullptr; + signature_I_PS stream_getInputPreset = nullptr; + signature_I_PS stream_getSessionId = nullptr; + + private: + AAudioLoader() {} + ~AAudioLoader(); + + // Load function pointers for specific signatures. + signature_I_PPB load_I_PPB(const char *name); + signature_CPH_I load_CPH_I(const char *name); + signature_V_PBI load_V_PBI(const char *name); + signature_V_PBPDPV load_V_PBPDPV(const char *name); + signature_V_PBPEPV load_V_PBPEPV(const char *name); + signature_I_PB load_I_PB(const char *name); + signature_I_PBPPS load_I_PBPPS(const char *name); + signature_I_PS load_I_PS(const char *name); + signature_L_PS load_L_PS(const char *name); + signature_F_PS load_F_PS(const char *name); + signature_B_PS load_B_PS(const char *name); + signature_I_PSI load_I_PSI(const char *name); + signature_I_PSPVIL load_I_PSPVIL(const char *name); + signature_I_PSCPVIL load_I_PSCPVIL(const char *name); + signature_I_PSTPTL load_I_PSTPTL(const char *name); + signature_I_PSKPLPL load_I_PSKPLPL(const char *name); + + void *mLibHandle = nullptr; +}; + +} // namespace oboe + +#endif //OBOE_AAUDIO_LOADER_H_ diff --git a/modules/juce_audio_devices/native/oboe/src/aaudio/AudioStreamAAudio.cpp b/modules/juce_audio_devices/native/oboe/src/aaudio/AudioStreamAAudio.cpp new file mode 100644 index 0000000000..33091ad5ad --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/aaudio/AudioStreamAAudio.cpp @@ -0,0 +1,620 @@ +/* + * Copyright 2016 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include +#include +#include + +#include "aaudio/AAudioLoader.h" +#include "aaudio/AudioStreamAAudio.h" +#include "common/AudioClock.h" +#include "common/OboeDebug.h" +#include "oboe/Utilities.h" + +#ifdef __ANDROID__ +#include +#endif + +#ifndef OBOE_FIX_FORCE_STARTING_TO_STARTED +// Workaround state problems in AAudio +// TODO Which versions does this occur in? Verify fixed in Q. +#define OBOE_FIX_FORCE_STARTING_TO_STARTED 1 +#endif // OBOE_FIX_FORCE_STARTING_TO_STARTED + +using namespace oboe; +AAudioLoader *AudioStreamAAudio::mLibLoader = nullptr; + +// 'C' wrapper for the data callback method +static aaudio_data_callback_result_t oboe_aaudio_data_callback_proc( + AAudioStream *stream, + void *userData, + void *audioData, + int32_t numFrames) { + + AudioStreamAAudio *oboeStream = reinterpret_cast(userData); + if (oboeStream != nullptr) { + return static_cast( + oboeStream->callOnAudioReady(stream, audioData, numFrames)); + + } else { + return static_cast(DataCallbackResult::Stop); + } +} + +// This runs in its own thread. +// Only one of these threads will be launched from internalErrorCallback(). +// It calls app error callbacks from a static function in case the stream gets deleted. +static void oboe_aaudio_error_thread_proc(AudioStreamAAudio *oboeStream, + Result error) { + LOGD("%s() - entering >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>", __func__); + oboeStream->requestStop(); + if (oboeStream->getCallback() != nullptr) { + oboeStream->getCallback()->onErrorBeforeClose(oboeStream, error); + } + oboeStream->close(); + if (oboeStream->getCallback() != nullptr) { + // Warning, oboeStream may get deleted by this callback. + oboeStream->getCallback()->onErrorAfterClose(oboeStream, error); + } + LOGD("%s() - exiting <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<", __func__); +} + +namespace oboe { + +/* + * Create a stream that uses Oboe Audio API. + */ +AudioStreamAAudio::AudioStreamAAudio(const AudioStreamBuilder &builder) + : AudioStream(builder) + , mAAudioStream(nullptr) { + mCallbackThreadEnabled.store(false); + LOGD("AudioStreamAAudio() call isSupported()"); + isSupported(); +} + +bool AudioStreamAAudio::isSupported() { + mLibLoader = AAudioLoader::getInstance(); + int openResult = mLibLoader->open(); + return openResult == 0; +} + +// Static 'C' wrapper for the error callback method. +// Launch a thread to handle the error. +// That other thread can safely stop, close and delete the stream. +void AudioStreamAAudio::internalErrorCallback( + AAudioStream *stream, + void *userData, + aaudio_result_t error) { + AudioStreamAAudio *oboeStream = reinterpret_cast(userData); + // These checks should be enough because we assume that the stream close() + // will join() any active callback threads and will not allow new callbacks. + if (oboeStream->wasErrorCallbackCalled()) { // block extra error callbacks + LOGE("%s() multiple error callbacks called!", __func__); + } else if (stream != oboeStream->getUnderlyingStream()) { + LOGD("%s() stream already closed", __func__); // can happen if there are bugs + } else { + // Handle error on a separate thread. + std::thread t(oboe_aaudio_error_thread_proc, oboeStream, + static_cast(error)); + t.detach(); + } +} + +void AudioStreamAAudio::logUnsupportedAttributes() { + int sdkVersion = getSdkVersion(); + + // These attributes are not supported pre Android "P" + if (sdkVersion < __ANDROID_API_P__) { + if (mUsage != Usage::Media) { + LOGW("Usage [AudioStreamBuilder::setUsage()] " + "is not supported on AAudio streams running on pre-Android P versions."); + } + + if (mContentType != ContentType::Music) { + LOGW("ContentType [AudioStreamBuilder::setContentType()] " + "is not supported on AAudio streams running on pre-Android P versions."); + } + + if (mSessionId != SessionId::None) { + LOGW("SessionId [AudioStreamBuilder::setSessionId()] " + "is not supported on AAudio streams running on pre-Android P versions."); + } + } +} + +Result AudioStreamAAudio::open() { + Result result = Result::OK; + + if (mAAudioStream != nullptr) { + return Result::ErrorInvalidState; + } + + result = AudioStream::open(); + if (result != Result::OK) { + return result; + } + + AAudioStreamBuilder *aaudioBuilder; + result = static_cast(mLibLoader->createStreamBuilder(&aaudioBuilder)); + if (result != Result::OK) { + return result; + } + + // Do not set INPUT capacity below 4096 because that prevents us from getting a FAST track + // when using the Legacy data path. + // If the app requests > 4096 then we allow it but we are less likely to get LowLatency. + // See internal bug b/80308183 for more details. + // Fixed in Q but let's still clip the capacity because high input capacity + // does not increase latency. + int32_t capacity = mBufferCapacityInFrames; + constexpr int kCapacityRequiredForFastLegacyTrack = 4096; // matches value in AudioFinger + if (mDirection == oboe::Direction::Input + && capacity != oboe::Unspecified + && capacity < kCapacityRequiredForFastLegacyTrack + && mPerformanceMode == oboe::PerformanceMode::LowLatency) { + capacity = kCapacityRequiredForFastLegacyTrack; + LOGD("AudioStreamAAudio.open() capacity changed from %d to %d for lower latency", + static_cast(mBufferCapacityInFrames), capacity); + } + mLibLoader->builder_setBufferCapacityInFrames(aaudioBuilder, capacity); + + mLibLoader->builder_setChannelCount(aaudioBuilder, mChannelCount); + mLibLoader->builder_setDeviceId(aaudioBuilder, mDeviceId); + mLibLoader->builder_setDirection(aaudioBuilder, static_cast(mDirection)); + mLibLoader->builder_setFormat(aaudioBuilder, static_cast(mFormat)); + mLibLoader->builder_setSampleRate(aaudioBuilder, mSampleRate); + mLibLoader->builder_setSharingMode(aaudioBuilder, + static_cast(mSharingMode)); + mLibLoader->builder_setPerformanceMode(aaudioBuilder, + static_cast(mPerformanceMode)); + + // These were added in P so we have to check for the function pointer. + if (mLibLoader->builder_setUsage != nullptr) { + mLibLoader->builder_setUsage(aaudioBuilder, + static_cast(mUsage)); + } + + if (mLibLoader->builder_setContentType != nullptr) { + mLibLoader->builder_setContentType(aaudioBuilder, + static_cast(mContentType)); + } + + if (mLibLoader->builder_setInputPreset != nullptr) { + mLibLoader->builder_setInputPreset(aaudioBuilder, + static_cast(mInputPreset)); + } + + if (mLibLoader->builder_setSessionId != nullptr) { + mLibLoader->builder_setSessionId(aaudioBuilder, + static_cast(mSessionId)); + } + + // TODO get more parameters from the builder? + + if (mStreamCallback != nullptr) { + mLibLoader->builder_setDataCallback(aaudioBuilder, oboe_aaudio_data_callback_proc, this); + mLibLoader->builder_setFramesPerDataCallback(aaudioBuilder, getFramesPerCallback()); + // If the data callback is not being used then the write method will return an error + // and the app can stop and close the stream. + mLibLoader->builder_setErrorCallback(aaudioBuilder, internalErrorCallback, this); + } + + // ============= OPEN THE STREAM ================ + { + AAudioStream *stream = nullptr; + result = static_cast(mLibLoader->builder_openStream(aaudioBuilder, &stream)); + mAAudioStream.store(stream); + } + if (result != Result::OK) { + goto error2; + } + + // Query and cache the stream properties + mDeviceId = mLibLoader->stream_getDeviceId(mAAudioStream); + mChannelCount = mLibLoader->stream_getChannelCount(mAAudioStream); + mSampleRate = mLibLoader->stream_getSampleRate(mAAudioStream); + mFormat = static_cast(mLibLoader->stream_getFormat(mAAudioStream)); + mSharingMode = static_cast(mLibLoader->stream_getSharingMode(mAAudioStream)); + mPerformanceMode = static_cast( + mLibLoader->stream_getPerformanceMode(mAAudioStream)); + mBufferCapacityInFrames = mLibLoader->stream_getBufferCapacity(mAAudioStream); + mBufferSizeInFrames = mLibLoader->stream_getBufferSize(mAAudioStream); + + + // These were added in P so we have to check for the function pointer. + if (mLibLoader->stream_getUsage != nullptr) { + mUsage = static_cast(mLibLoader->stream_getUsage(mAAudioStream)); + } + if (mLibLoader->stream_getContentType != nullptr) { + mContentType = static_cast(mLibLoader->stream_getContentType(mAAudioStream)); + } + if (mLibLoader->stream_getInputPreset != nullptr) { + mInputPreset = static_cast(mLibLoader->stream_getInputPreset(mAAudioStream)); + } + if (mLibLoader->stream_getSessionId != nullptr) { + mSessionId = static_cast(mLibLoader->stream_getSessionId(mAAudioStream)); + } else { + mSessionId = SessionId::None; + } + + LOGD("AudioStreamAAudio.open() app format = %d", static_cast(mFormat)); + LOGD("AudioStreamAAudio.open() sample rate = %d", static_cast(mSampleRate)); + LOGD("AudioStreamAAudio.open() capacity = %d", static_cast(mBufferCapacityInFrames)); + +error2: + mLibLoader->builder_delete(aaudioBuilder); + LOGD("AudioStreamAAudio.open: AAudioStream_Open() returned %s, mAAudioStream = %p", + mLibLoader->convertResultToText(static_cast(result)), + mAAudioStream.load()); + return result; +} + +Result AudioStreamAAudio::close() { + // The main reason we have this mutex if to prevent a collision between a call + // by the application to stop a stream at the same time that an onError callback + // is being executed because of a disconnect. The close will delete the stream, + // which could otherwise cause the requestStop() to crash. + std::lock_guard lock(mLock); + + AudioStream::close(); + + // This will delete the AAudio stream object so we need to null out the pointer. + AAudioStream *stream = mAAudioStream.exchange(nullptr); + if (stream != nullptr) { + return static_cast(mLibLoader->stream_close(stream)); + } else { + return Result::ErrorClosed; + } +} + +DataCallbackResult AudioStreamAAudio::callOnAudioReady(AAudioStream *stream, + void *audioData, + int32_t numFrames) { + DataCallbackResult result = fireDataCallback(audioData, numFrames); + if (result == DataCallbackResult::Continue) { + return result; + } else { + if (result == DataCallbackResult::Stop) { + LOGE("Oboe callback returned DataCallbackResult::Stop"); + } else { + LOGE("Oboe callback returned unexpected value = %d", result); + } + + if (getSdkVersion() <= __ANDROID_API_P__) { + launchStopThread(); + if (isMMapUsed()) { + return DataCallbackResult::Stop; + } else { + // Legacy stream <= API_P cannot be restarted after returning Stop. + return DataCallbackResult::Continue; + } + } else { + return DataCallbackResult::Stop; // OK >= API_Q + } + } +} + +Result AudioStreamAAudio::requestStart() { + std::lock_guard lock(mLock); + AAudioStream *stream = mAAudioStream.load(); + if (stream != nullptr) { + // Avoid state machine errors in O_MR1. + if (getSdkVersion() <= __ANDROID_API_O_MR1__) { + StreamState state = static_cast(mLibLoader->stream_getState(stream)); + if (state == StreamState::Starting || state == StreamState::Started) { + // WARNING: On P, AAudio is returning ErrorInvalidState for Output and OK for Input. + return Result::OK; + } + } + if (mStreamCallback != nullptr) { // Was a callback requested? + setDataCallbackEnabled(true); + } + return static_cast(mLibLoader->stream_requestStart(stream)); + } else { + return Result::ErrorClosed; + } +} + +Result AudioStreamAAudio::requestPause() { + std::lock_guard lock(mLock); + AAudioStream *stream = mAAudioStream.load(); + if (stream != nullptr) { + // Avoid state machine errors in O_MR1. + if (getSdkVersion() <= __ANDROID_API_O_MR1__) { + StreamState state = static_cast(mLibLoader->stream_getState(stream)); + if (state == StreamState::Pausing || state == StreamState::Paused) { + return Result::OK; + } + } + return static_cast(mLibLoader->stream_requestPause(stream)); + } else { + return Result::ErrorClosed; + } +} + +Result AudioStreamAAudio::requestFlush() { + std::lock_guard lock(mLock); + AAudioStream *stream = mAAudioStream.load(); + if (stream != nullptr) { + // Avoid state machine errors in O_MR1. + if (getSdkVersion() <= __ANDROID_API_O_MR1__) { + StreamState state = static_cast(mLibLoader->stream_getState(stream)); + if (state == StreamState::Flushing || state == StreamState::Flushed) { + return Result::OK; + } + } + return static_cast(mLibLoader->stream_requestFlush(stream)); + } else { + return Result::ErrorClosed; + } +} + +Result AudioStreamAAudio::requestStop() { + std::lock_guard lock(mLock); + AAudioStream *stream = mAAudioStream.load(); + if (stream != nullptr) { + // Avoid state machine errors in O_MR1. + if (getSdkVersion() <= __ANDROID_API_O_MR1__) { + StreamState state = static_cast(mLibLoader->stream_getState(stream)); + if (state == StreamState::Stopping || state == StreamState::Stopped) { + return Result::OK; + } + } + return static_cast(mLibLoader->stream_requestStop(stream)); + } else { + return Result::ErrorClosed; + } +} + +ResultWithValue AudioStreamAAudio::write(const void *buffer, + int32_t numFrames, + int64_t timeoutNanoseconds) { + AAudioStream *stream = mAAudioStream.load(); + if (stream != nullptr) { + int32_t result = mLibLoader->stream_write(mAAudioStream, buffer, + numFrames, timeoutNanoseconds); + return ResultWithValue::createBasedOnSign(result); + } else { + return ResultWithValue(Result::ErrorClosed); + } +} + +ResultWithValue AudioStreamAAudio::read(void *buffer, + int32_t numFrames, + int64_t timeoutNanoseconds) { + AAudioStream *stream = mAAudioStream.load(); + if (stream != nullptr) { + int32_t result = mLibLoader->stream_read(mAAudioStream, buffer, + numFrames, timeoutNanoseconds); + return ResultWithValue::createBasedOnSign(result); + } else { + return ResultWithValue(Result::ErrorClosed); + } +} + + +// AAudioStream_waitForStateChange() can crash if it is waiting on a stream and that stream +// is closed from another thread. We do not want to lock the stream for the duration of the call. +// So we call AAudioStream_waitForStateChange() with a timeout of zero so that it will not block. +// Then we can do our own sleep with the lock unlocked. +Result AudioStreamAAudio::waitForStateChange(StreamState currentState, + StreamState *nextState, + int64_t timeoutNanoseconds) { + Result oboeResult = Result::ErrorTimeout; + int64_t sleepTimeNanos = 20 * kNanosPerMillisecond; // arbitrary + aaudio_stream_state_t currentAAudioState = static_cast(currentState); + + aaudio_result_t result = AAUDIO_OK; + int64_t timeLeftNanos = timeoutNanoseconds; + + mLock.lock(); + while (true) { + // Do we still have an AAudio stream? If not then stream must have been closed. + AAudioStream *stream = mAAudioStream.load(); + if (stream == nullptr) { + if (nextState != nullptr) { + *nextState = StreamState::Closed; + } + oboeResult = Result::ErrorClosed; + break; + } + + // Update and query state change with no blocking. + aaudio_stream_state_t aaudioNextState; + result = mLibLoader->stream_waitForStateChange( + mAAudioStream, + currentAAudioState, + &aaudioNextState, + 0); // timeout=0 for non-blocking + // AAudio will return AAUDIO_ERROR_TIMEOUT if timeout=0 and the state does not change. + if (result != AAUDIO_OK && result != AAUDIO_ERROR_TIMEOUT) { + oboeResult = static_cast(result); + break; + } +#if OBOE_FIX_FORCE_STARTING_TO_STARTED + if (aaudioNextState == static_cast(StreamState::Starting)) { + aaudioNextState = static_cast(StreamState::Started); + } +#endif // OBOE_FIX_FORCE_STARTING_TO_STARTED + if (nextState != nullptr) { + *nextState = static_cast(aaudioNextState); + } + if (currentAAudioState != aaudioNextState) { // state changed? + oboeResult = Result::OK; + break; + } + + // Did we timeout or did user ask for non-blocking? + if (timeLeftNanos <= 0) { + break; + } + + // No change yet so sleep. + mLock.unlock(); // Don't sleep while locked. + if (sleepTimeNanos > timeLeftNanos) { + sleepTimeNanos = timeLeftNanos; // last little bit + } + AudioClock::sleepForNanos(sleepTimeNanos); + timeLeftNanos -= sleepTimeNanos; + mLock.lock(); + } + + mLock.unlock(); + return oboeResult; +} + +ResultWithValue AudioStreamAAudio::setBufferSizeInFrames(int32_t requestedFrames) { + + AAudioStream *stream = mAAudioStream.load(); + + if (stream != nullptr) { + + if (requestedFrames > mBufferCapacityInFrames) { + requestedFrames = mBufferCapacityInFrames; + } + int32_t newBufferSize = mLibLoader->stream_setBufferSize(mAAudioStream, requestedFrames); + + // Cache the result if it's valid + if (newBufferSize > 0) mBufferSizeInFrames = newBufferSize; + + return ResultWithValue::createBasedOnSign(newBufferSize); + + } else { + return ResultWithValue(Result::ErrorClosed); + } +} + +StreamState AudioStreamAAudio::getState() const { + AAudioStream *stream = mAAudioStream.load(); + if (stream != nullptr) { + aaudio_stream_state_t aaudioState = mLibLoader->stream_getState(stream); +#if OBOE_FIX_FORCE_STARTING_TO_STARTED + if (aaudioState == AAUDIO_STREAM_STATE_STARTING) { + aaudioState = AAUDIO_STREAM_STATE_STARTED; + } +#endif // OBOE_FIX_FORCE_STARTING_TO_STARTED + return static_cast(aaudioState); + } else { + return StreamState::Closed; + } +} + +int32_t AudioStreamAAudio::getBufferSizeInFrames() { + AAudioStream *stream = mAAudioStream.load(); + if (stream != nullptr) { + mBufferSizeInFrames = mLibLoader->stream_getBufferSize(stream); + } + return mBufferSizeInFrames; +} + +int32_t AudioStreamAAudio::getFramesPerBurst() { + AAudioStream *stream = mAAudioStream.load(); + if (stream != nullptr) { + mFramesPerBurst = mLibLoader->stream_getFramesPerBurst(stream); + } + return mFramesPerBurst; +} + +void AudioStreamAAudio::updateFramesRead() { + AAudioStream *stream = mAAudioStream.load(); + if (stream != nullptr) { + mFramesRead = mLibLoader->stream_getFramesRead(stream); + } +} + +void AudioStreamAAudio::updateFramesWritten() { + AAudioStream *stream = mAAudioStream.load(); + if (stream != nullptr) { + mFramesWritten = mLibLoader->stream_getFramesWritten(stream); + } +} + +ResultWithValue AudioStreamAAudio::getXRunCount() const { + AAudioStream *stream = mAAudioStream.load(); + if (stream != nullptr) { + return ResultWithValue::createBasedOnSign(mLibLoader->stream_getXRunCount(stream)); + } else { + return ResultWithValue(Result::ErrorNull); + } +} + +Result AudioStreamAAudio::getTimestamp(clockid_t clockId, + int64_t *framePosition, + int64_t *timeNanoseconds) { + AAudioStream *stream = mAAudioStream.load(); + if (stream != nullptr) { + if (getState() != StreamState::Started) { + return Result::ErrorInvalidState; + } + return static_cast(mLibLoader->stream_getTimestamp(stream, clockId, + framePosition, timeNanoseconds)); + } else { + return Result::ErrorNull; + } +} + +ResultWithValue AudioStreamAAudio::calculateLatencyMillis() { + AAudioStream *stream = mAAudioStream.load(); + if (stream == nullptr) { + return ResultWithValue(Result::ErrorClosed); + } + + // Get the time that a known audio frame was presented. + int64_t hardwareFrameIndex; + int64_t hardwareFrameHardwareTime; + auto result = getTimestamp(CLOCK_MONOTONIC, + &hardwareFrameIndex, + &hardwareFrameHardwareTime); + if (result != oboe::Result::OK) { + return ResultWithValue(static_cast(result)); + } + + // Get counter closest to the app. + bool isOutput = (getDirection() == oboe::Direction::Output); + int64_t appFrameIndex = isOutput ? getFramesWritten() : getFramesRead(); + + // Assume that the next frame will be processed at the current time + using namespace std::chrono; + int64_t appFrameAppTime = + duration_cast(steady_clock::now().time_since_epoch()).count(); + + // Calculate the number of frames between app and hardware + int64_t frameIndexDelta = appFrameIndex - hardwareFrameIndex; + + // Calculate the time which the next frame will be or was presented + int64_t frameTimeDelta = (frameIndexDelta * oboe::kNanosPerSecond) / getSampleRate(); + int64_t appFrameHardwareTime = hardwareFrameHardwareTime + frameTimeDelta; + + // The current latency is the difference in time between when the current frame is at + // the app and when it is at the hardware. + double latencyNanos = static_cast(isOutput + ? (appFrameHardwareTime - appFrameAppTime) // hardware is later + : (appFrameAppTime - appFrameHardwareTime)); // hardware is earlier + double latencyMillis = latencyNanos / kNanosPerMillisecond; + + return ResultWithValue(latencyMillis); +} + +bool AudioStreamAAudio::isMMapUsed() { + AAudioStream *stream = mAAudioStream.load(); + if (stream != nullptr) { + return mLibLoader->stream_isMMapUsed(stream); + } else { + return false; + } +} + +} // namespace oboe diff --git a/modules/juce_audio_devices/native/oboe/src/aaudio/AudioStreamAAudio.h b/modules/juce_audio_devices/native/oboe/src/aaudio/AudioStreamAAudio.h new file mode 100644 index 0000000000..2a04396271 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/aaudio/AudioStreamAAudio.h @@ -0,0 +1,123 @@ +/* + * Copyright 2016 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_STREAM_AAUDIO_H_ +#define OBOE_STREAM_AAUDIO_H_ + +#include +#include +#include + +#include "oboe/AudioStreamBuilder.h" +#include "oboe/AudioStream.h" +#include "oboe/Definitions.h" +#include "AAudioLoader.h" + +namespace oboe { + +/** + * Implementation of OboeStream that uses AAudio. + * + * Do not create this class directly. + * Use an OboeStreamBuilder to create one. + */ +class AudioStreamAAudio : public AudioStream { +public: + AudioStreamAAudio(); + explicit AudioStreamAAudio(const AudioStreamBuilder &builder); + + virtual ~AudioStreamAAudio() = default; + + /** + * + * @return true if AAudio is supported on this device. + */ + static bool isSupported(); + + // These functions override methods in AudioStream. + // See AudioStream for documentation. + Result open() override; + Result close() override; + + Result requestStart() override; + Result requestPause() override; + Result requestFlush() override; + Result requestStop() override; + + ResultWithValue write(const void *buffer, + int32_t numFrames, + int64_t timeoutNanoseconds) override; + + ResultWithValue read(void *buffer, + int32_t numFrames, + int64_t timeoutNanoseconds) override; + + ResultWithValue setBufferSizeInFrames(int32_t requestedFrames) override; + int32_t getBufferSizeInFrames() override; + int32_t getFramesPerBurst() override; + ResultWithValue getXRunCount() const override; + bool isXRunCountSupported() const override { return true; } + + ResultWithValue calculateLatencyMillis() override; + + Result waitForStateChange(StreamState currentState, + StreamState *nextState, + int64_t timeoutNanoseconds) override; + + Result getTimestamp(clockid_t clockId, + int64_t *framePosition, + int64_t *timeNanoseconds) override; + + StreamState getState() const override; + + AudioApi getAudioApi() const override { + return AudioApi::AAudio; + } + + DataCallbackResult callOnAudioReady(AAudioStream *stream, + void *audioData, + int32_t numFrames); + +protected: + static void internalErrorCallback( + AAudioStream *stream, + void *userData, + aaudio_result_t error); + + void *getUnderlyingStream() const override { + return mAAudioStream.load(); + } + + void updateFramesRead() override; + void updateFramesWritten() override; + + void logUnsupportedAttributes(); + +private: + + bool isMMapUsed(); + + std::atomic mCallbackThreadEnabled; + + // pointer to the underlying AAudio stream, valid if open, null if closed + std::atomic mAAudioStream{nullptr}; + + static AAudioLoader *mLibLoader; +}; + +} // namespace oboe + +#endif // OBOE_STREAM_AAUDIO_H_ diff --git a/modules/juce_audio_devices/native/oboe/src/common/AudioClock.h b/modules/juce_audio_devices/native/oboe/src/common/AudioClock.h new file mode 100644 index 0000000000..3fe20cb0a6 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/common/AudioClock.h @@ -0,0 +1,75 @@ +/* + * Copyright (C) 2016 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_AUDIO_CLOCK_H +#define OBOE_AUDIO_CLOCK_H + +#include +#include +#include "oboe/Definitions.h" + +namespace oboe { + +// TODO: Move this class into the public headers because it is useful when calculating stream latency +class AudioClock { +public: + static int64_t getNanoseconds(clockid_t clockId = CLOCK_MONOTONIC) { + struct timespec time; + int result = clock_gettime(clockId, &time); + if (result < 0) { + return result; + } + return (time.tv_sec * kNanosPerSecond) + time.tv_nsec; + } + + /** + * Sleep until the specified time. + * + * @param nanoTime time to wake up + * @param clockId CLOCK_MONOTONIC is default + * @return 0 or a negative error, eg. -EINTR + */ + + static int sleepUntilNanoTime(int64_t nanoTime, clockid_t clockId = CLOCK_MONOTONIC) { + struct timespec time; + time.tv_sec = nanoTime / kNanosPerSecond; + time.tv_nsec = nanoTime - (time.tv_sec * kNanosPerSecond); + return 0 - clock_nanosleep(clockId, TIMER_ABSTIME, &time, NULL); + } + + /** + * Sleep for the specified number of nanoseconds in real-time. + * Return immediately with 0 if a negative nanoseconds is specified. + * + * @param nanoseconds time to sleep + * @param clockId CLOCK_REALTIME is default + * @return 0 or a negative error, eg. -EINTR + */ + + static int sleepForNanos(int64_t nanoseconds, clockid_t clockId = CLOCK_REALTIME) { + if (nanoseconds > 0) { + struct timespec time; + time.tv_sec = nanoseconds / kNanosPerSecond; + time.tv_nsec = nanoseconds - (time.tv_sec * kNanosPerSecond); + return 0 - clock_nanosleep(clockId, 0, &time, NULL); + } + return 0; + } +}; + +} // namespace oboe + +#endif //OBOE_AUDIO_CLOCK_H diff --git a/modules/juce_audio_devices/native/oboe/src/common/AudioSourceCaller.cpp b/modules/juce_audio_devices/native/oboe/src/common/AudioSourceCaller.cpp new file mode 100644 index 0000000000..0180c2264c --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/common/AudioSourceCaller.cpp @@ -0,0 +1,38 @@ +/* + * Copyright 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include "AudioSourceCaller.h" + +using namespace oboe; +using namespace flowgraph; + +int32_t AudioSourceCaller::onProcessFixedBlock(uint8_t *buffer, int32_t numBytes) { + oboe::AudioStreamCallback *callback = mStream->getCallback(); + int32_t result = 0; + int32_t numFrames = numBytes / mStream->getBytesPerFrame(); + if (callback != nullptr) { + DataCallbackResult callbackResult = callback->onAudioReady(mStream, buffer, numFrames); + // onAudioReady() does not return the number of bytes processed so we have to assume all. + result = (callbackResult == DataCallbackResult::Continue) + ? numBytes + : -1; + } else { + auto readResult = mStream->read(buffer, numFrames, mTimeoutNanos); + if (!readResult) return (int32_t) readResult.error(); + result = readResult.value() * mStream->getBytesPerFrame(); + } + return result; +} diff --git a/modules/juce_audio_devices/native/oboe/src/common/AudioSourceCaller.h b/modules/juce_audio_devices/native/oboe/src/common/AudioSourceCaller.h new file mode 100644 index 0000000000..d196d410ea --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/common/AudioSourceCaller.h @@ -0,0 +1,83 @@ +/* + * Copyright 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_AUDIO_SOURCE_CALLER_H +#define OBOE_AUDIO_SOURCE_CALLER_H + +#include "OboeDebug.h" +#include "oboe/Oboe.h" + +#include "flowgraph/FlowGraphNode.h" +#include "FixedBlockReader.h" + +namespace oboe { + +class AudioStreamCallback; +class AudioStream; + +/** + * For output streams that use a callback, call the application for more data. + * For input streams that do not use a callback, read from the stream. + */ +class AudioSourceCaller : public flowgraph::FlowGraphSource, public FixedBlockProcessor { +public: + AudioSourceCaller(int32_t channelCount, int32_t framesPerCallback, int32_t bytesPerSample) + : FlowGraphSource(channelCount) + , mBlockReader(*this) { + mBlockReader.open(channelCount * framesPerCallback * bytesPerSample); + } + + /** + * Set the stream to use as a source of data. + * @param stream + */ + void setStream(oboe::AudioStream *stream) { + mStream = stream; + } + + oboe::AudioStream *getStream() { + return mStream; + } + + /** + * Timeout value to use when calling audioStream->read(). + * @param timeoutNanos Zero for no timeout or time in nanoseconds. + */ + void setTimeoutNanos(int64_t timeoutNanos) { + mTimeoutNanos = timeoutNanos; + } + + int64_t getTimeoutNanos() const { + return mTimeoutNanos; + } + + /** + * Called internally for block size adaptation. + * @param buffer + * @param numBytes + * @return + */ + int32_t onProcessFixedBlock(uint8_t *buffer, int32_t numBytes) override; + +protected: + oboe::AudioStream *mStream = nullptr; + int64_t mTimeoutNanos = 0; + + FixedBlockReader mBlockReader; +}; + +} +#endif //OBOE_AUDIO_SOURCE_CALLER_H diff --git a/modules/juce_audio_devices/native/oboe/src/common/AudioStream.cpp b/modules/juce_audio_devices/native/oboe/src/common/AudioStream.cpp new file mode 100644 index 0000000000..f0fd2f238a --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/common/AudioStream.cpp @@ -0,0 +1,211 @@ +/* + * Copyright 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include +#include +#include + +#include +#include "OboeDebug.h" +#include "AudioClock.h" +#include + +namespace oboe { + +/* + * AudioStream + */ +AudioStream::AudioStream(const AudioStreamBuilder &builder) + : AudioStreamBase(builder) { +} + +Result AudioStream::close() { + // Update local counters so they can be read after the close. + updateFramesWritten(); + updateFramesRead(); + return Result::OK; +} + +DataCallbackResult AudioStream::fireDataCallback(void *audioData, int32_t numFrames) { + if (!isDataCallbackEnabled()) { + LOGW("AudioStream::%s() called with data callback disabled!", __func__); + return DataCallbackResult::Stop; // We should not be getting called any more. + } + + // TODO remove + int scheduler = sched_getscheduler(0) & ~SCHED_RESET_ON_FORK; // for current thread + if (scheduler != mPreviousScheduler) { + LOGD("AudioStream::%s() scheduler = %s", __func__, + ((scheduler == SCHED_FIFO) ? "SCHED_FIFO" : + ((scheduler == SCHED_OTHER) ? "SCHED_OTHER" : + ((scheduler == SCHED_RR) ? "SCHED_RR" : "UNKNOWN"))) + ); + mPreviousScheduler = scheduler; + } + + DataCallbackResult result; + if (mStreamCallback == nullptr) { + result = onDefaultCallback(audioData, numFrames); + } else { + result = mStreamCallback->onAudioReady(this, audioData, numFrames); + } + // On Oreo, we might get called after returning stop. + // So block that here. + setDataCallbackEnabled(result == DataCallbackResult::Continue); + + return result; +} + +Result AudioStream::waitForStateTransition(StreamState startingState, + StreamState endingState, + int64_t timeoutNanoseconds) +{ + StreamState state; + { + std::lock_guard lock(mLock); + state = getState(); + if (state == StreamState::Closed) { + return Result::ErrorClosed; + } else if (state == StreamState::Disconnected) { + return Result::ErrorDisconnected; + } + } + + StreamState nextState = state; + // TODO Should this be a while()?! + if (state == startingState && state != endingState) { + Result result = waitForStateChange(state, &nextState, timeoutNanoseconds); + if (result != Result::OK) { + return result; + } + } + + if (nextState != endingState) { + return Result::ErrorInvalidState; + } else { + return Result::OK; + } +} + +Result AudioStream::start(int64_t timeoutNanoseconds) +{ + Result result = requestStart(); + if (result != Result::OK) return result; + if (timeoutNanoseconds <= 0) return result; + return waitForStateTransition(StreamState::Starting, + StreamState::Started, timeoutNanoseconds); +} + +Result AudioStream::pause(int64_t timeoutNanoseconds) +{ + Result result = requestPause(); + if (result != Result::OK) return result; + if (timeoutNanoseconds <= 0) return result; + return waitForStateTransition(StreamState::Pausing, + StreamState::Paused, timeoutNanoseconds); +} + +Result AudioStream::flush(int64_t timeoutNanoseconds) +{ + Result result = requestFlush(); + if (result != Result::OK) return result; + if (timeoutNanoseconds <= 0) return result; + return waitForStateTransition(StreamState::Flushing, + StreamState::Flushed, timeoutNanoseconds); +} + +Result AudioStream::stop(int64_t timeoutNanoseconds) +{ + Result result = requestStop(); + if (result != Result::OK) return result; + if (timeoutNanoseconds <= 0) return result; + return waitForStateTransition(StreamState::Stopping, + StreamState::Stopped, timeoutNanoseconds); +} + +int32_t AudioStream::getBytesPerSample() const { + return convertFormatToSizeInBytes(mFormat); +} + +int64_t AudioStream::getFramesRead() { + updateFramesRead(); + return mFramesRead; +} + +int64_t AudioStream::getFramesWritten() { + updateFramesWritten(); + return mFramesWritten; +} + +ResultWithValue AudioStream::getAvailableFrames() { + int64_t readCounter = getFramesRead(); + if (readCounter < 0) return ResultWithValue::createBasedOnSign(readCounter); + int64_t writeCounter = getFramesWritten(); + if (writeCounter < 0) return ResultWithValue::createBasedOnSign(writeCounter); + int32_t framesAvailable = writeCounter - readCounter; + return ResultWithValue(framesAvailable); +} + +ResultWithValue AudioStream::waitForAvailableFrames(int32_t numFrames, + int64_t timeoutNanoseconds) { + if (numFrames == 0) return Result::OK; + if (numFrames < 0) return Result::ErrorOutOfRange; + + int64_t framesAvailable = 0; + int64_t burstInNanos = getFramesPerBurst() * kNanosPerSecond / getSampleRate(); + bool ready = false; + int64_t deadline = AudioClock::getNanoseconds() + timeoutNanoseconds; + do { + ResultWithValue result = getAvailableFrames(); + if (!result) return result; + framesAvailable = result.value(); + ready = (framesAvailable >= numFrames); + if (!ready) { + int64_t now = AudioClock::getNanoseconds(); + if (now > deadline) break; + AudioClock::sleepForNanos(burstInNanos); + } + } while (!ready); + return (!ready) + ? ResultWithValue(Result::ErrorTimeout) + : ResultWithValue(framesAvailable); +} + +ResultWithValue AudioStream::getTimestamp(clockid_t clockId) { + FrameTimestamp frame; + Result result = getTimestamp(clockId, &frame.position, &frame.timestamp); + if (result == Result::OK){ + return ResultWithValue(frame); + } else { + return ResultWithValue(static_cast(result)); + } +} + +static void oboe_stop_thread_proc(AudioStream *oboeStream) { + LOGD("%s() called ----)))))", __func__); + if (oboeStream != nullptr) { + oboeStream->requestStop(); + } + LOGD("%s() returning (((((----", __func__); +} + +void AudioStream::launchStopThread() { + // Stop this stream on a separate thread + std::thread t(oboe_stop_thread_proc, this); + t.detach(); +} + +} // namespace oboe diff --git a/modules/juce_audio_devices/native/oboe/src/common/AudioStreamBuilder.cpp b/modules/juce_audio_devices/native/oboe/src/common/AudioStreamBuilder.cpp new file mode 100644 index 0000000000..401beea279 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/common/AudioStreamBuilder.cpp @@ -0,0 +1,187 @@ +/* + * Copyright 2016 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include + +#include "aaudio/AudioStreamAAudio.h" +#include "FilterAudioStream.h" +#include "OboeDebug.h" +#include "oboe/Oboe.h" +#include "oboe/AudioStreamBuilder.h" +#include "opensles/AudioInputStreamOpenSLES.h" +#include "opensles/AudioOutputStreamOpenSLES.h" +#include "opensles/AudioStreamOpenSLES.h" +#include "QuirksManager.h" + +namespace oboe { + +/** + * The following default values are used when oboe does not have any better way of determining the optimal values + * for an audio stream. This can happen when: + * + * - Client is creating a stream on API < 26 (OpenSLES) but has not supplied the optimal sample + * rate and/or frames per burst + * - Client is creating a stream on API 16 (OpenSLES) where AudioManager.PROPERTY_OUTPUT_* values + * are not available + */ +int32_t DefaultStreamValues::SampleRate = 48000; // Common rate for mobile audio and video +int32_t DefaultStreamValues::FramesPerBurst = 192; // 4 msec at 48000 Hz +int32_t DefaultStreamValues::ChannelCount = 2; // Stereo + +constexpr int kBufferSizeInBurstsForLowLatencyStreams = 2; + +#ifndef OBOE_ENABLE_AAUDIO +// Set OBOE_ENABLE_AAUDIO to 0 if you want to disable the AAudio API. +// This might be useful if you want to force all the unit tests to use OpenSL ES. +#define OBOE_ENABLE_AAUDIO 1 +#endif + +bool AudioStreamBuilder::isAAudioSupported() { + return AudioStreamAAudio::isSupported() && OBOE_ENABLE_AAUDIO; +} + +bool AudioStreamBuilder::isAAudioRecommended() { + // See https://github.com/google/oboe/issues/40, + // AAudio may not be stable on Android O, depending on how it is used. + // To be safe, use AAudio only on O_MR1 and above. + return (getSdkVersion() >= __ANDROID_API_O_MR1__) && isAAudioSupported(); +} + +AudioStream *AudioStreamBuilder::build() { + AudioStream *stream = nullptr; + if (isAAudioRecommended() && mAudioApi != AudioApi::OpenSLES) { + stream = new AudioStreamAAudio(*this); + } else if (isAAudioSupported() && mAudioApi == AudioApi::AAudio) { + stream = new AudioStreamAAudio(*this); + LOGE("Creating AAudio stream on 8.0 because it was specified. This is error prone."); + } else { + if (getDirection() == oboe::Direction::Output) { + stream = new AudioOutputStreamOpenSLES(*this); + } else if (getDirection() == oboe::Direction::Input) { + stream = new AudioInputStreamOpenSLES(*this); + } + } + return stream; +} + +bool AudioStreamBuilder::isCompatible(AudioStreamBase &other) { + return getSampleRate() == other.getSampleRate() + && getFormat() == other.getFormat() + && getChannelCount() == other.getChannelCount(); +} + +Result AudioStreamBuilder::openStream(AudioStream **streamPP) { + Result result = Result::OK; + LOGD("%s() %s -------- %s --------", + __func__, getDirection() == Direction::Input ? "INPUT" : "OUTPUT", getVersionText()); + + if (streamPP == nullptr) { + return Result::ErrorNull; + } + *streamPP = nullptr; + + AudioStream *streamP = nullptr; + + // Maybe make a FilterInputStream. + AudioStreamBuilder childBuilder(*this); + // Check need for conversion and modify childBuilder for optimal stream. + bool conversionNeeded = QuirksManager::getInstance().isConversionNeeded(*this, childBuilder); + // Do we need to make a child stream and convert. + if (conversionNeeded) { + AudioStream *tempStream; + + result = childBuilder.openStream(&tempStream); + if (result != Result::OK) { + return result; + } + + if (isCompatible(*tempStream)) { + // Everything matches so we can just use the child stream directly. + *streamPP = tempStream; + return result; + } else { + AudioStreamBuilder parentBuilder = *this; + // Build a stream that is as close as possible to the childStream. + if (getFormat() == oboe::AudioFormat::Unspecified) { + parentBuilder.setFormat(tempStream->getFormat()); + } + if (getChannelCount() == oboe::Unspecified) { + parentBuilder.setChannelCount(tempStream->getChannelCount()); + } + if (getSampleRate() == oboe::Unspecified) { + parentBuilder.setSampleRate(tempStream->getSampleRate()); + } + + // Use childStream in a FilterAudioStream. + LOGD("%s() create a FilterAudioStream for data conversion.", __func__); + FilterAudioStream *filterStream = new FilterAudioStream(parentBuilder, tempStream); + result = filterStream->configureFlowGraph(); + if (result != Result::OK) { + filterStream->close(); + delete filterStream; + // Just open streamP the old way. + } else { + streamP = static_cast(filterStream); + } + } + } + + if (streamP == nullptr) { + streamP = build(); + if (streamP == nullptr) { + return Result::ErrorNull; + } + } + + result = streamP->open(); // TODO review API + if (result == Result::OK) { + + int32_t optimalBufferSize = -1; + // Use a reasonable default buffer size. + if (streamP->getDirection() == Direction::Input) { + // For input, small size does not improve latency because the stream is usually + // run close to empty. And a low size can result in XRuns so always use the maximum. + optimalBufferSize = streamP->getBufferCapacityInFrames(); + } else if (streamP->getPerformanceMode() == PerformanceMode::LowLatency + && streamP->getDirection() == Direction::Output) { // Output check is redundant. + optimalBufferSize = streamP->getFramesPerBurst() * + kBufferSizeInBurstsForLowLatencyStreams; + } + if (optimalBufferSize >= 0) { + auto setBufferResult = streamP->setBufferSizeInFrames(optimalBufferSize); + if (!setBufferResult) { + LOGW("Failed to setBufferSizeInFrames(%d). Error was %s", + optimalBufferSize, + convertToText(setBufferResult.error())); + } + } + + *streamPP = streamP; + } else { + delete streamP; + } + return result; +} + +Result AudioStreamBuilder::openManagedStream(oboe::ManagedStream &stream) { + stream.reset(); + AudioStream *streamptr; + auto result = openStream(&streamptr); + stream.reset(streamptr); + return result; +} + +} // namespace oboe \ No newline at end of file diff --git a/modules/juce_audio_devices/native/oboe/src/common/DataConversionFlowGraph.cpp b/modules/juce_audio_devices/native/oboe/src/common/DataConversionFlowGraph.cpp new file mode 100644 index 0000000000..98536a9a61 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/common/DataConversionFlowGraph.cpp @@ -0,0 +1,210 @@ +/* + * Copyright (C) 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include + +#include "OboeDebug.h" +#include "DataConversionFlowGraph.h" +#include "SourceFloatCaller.h" +#include "SourceI16Caller.h" + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +using namespace oboe; +using namespace flowgraph; +using namespace resampler; + +void DataConversionFlowGraph::setSource(const void *buffer, int32_t numFrames) { + mSource->setData(buffer, numFrames); +} + +static MultiChannelResampler::Quality convertOboeSRQualityToMCR(SampleRateConversionQuality quality) { + switch (quality) { + case SampleRateConversionQuality::Fastest: + return MultiChannelResampler::Quality::Fastest; + case SampleRateConversionQuality::Low: + return MultiChannelResampler::Quality::Low; + default: + case SampleRateConversionQuality::Medium: + return MultiChannelResampler::Quality::Medium; + case SampleRateConversionQuality::High: + return MultiChannelResampler::Quality::High; + case SampleRateConversionQuality::Best: + return MultiChannelResampler::Quality::Best; + } +} + +// Chain together multiple processors. +// Callback Output +// Use SourceCaller that calls original app callback from the flowgraph. +// The child callback from FilteredAudioStream read()s from the flowgraph. +// Callback Input +// Child callback from FilteredAudioStream writes()s to the flowgraph. +// The output of the flowgraph goes through a BlockWriter to the app callback. +// Blocking Write +// Write buffer is set on an AudioSource. +// Data is pulled through the graph and written to the child stream. +// Blocking Read +// Reads in a loop from the flowgraph Sink to fill the read buffer. +// A SourceCaller then does a blocking read from the child Stream. +// +Result DataConversionFlowGraph::configure(AudioStream *sourceStream, AudioStream *sinkStream) { + + FlowGraphPortFloatOutput *lastOutput = nullptr; + + bool isOutput = sourceStream->getDirection() == Direction::Output; + mFilterStream = isOutput ? sourceStream : sinkStream; + + AudioFormat sourceFormat = sourceStream->getFormat(); + int32_t sourceChannelCount = sourceStream->getChannelCount(); + int32_t sourceSampleRate = sourceStream->getSampleRate(); + + AudioFormat sinkFormat = sinkStream->getFormat(); + int32_t sinkChannelCount = sinkStream->getChannelCount(); + int32_t sinkSampleRate = sinkStream->getSampleRate(); + + LOGD("%s() flowgraph converts channels: %d to %d, format: %d to %d, rate: %d to %d, qual = %d", + __func__, + sourceChannelCount, sinkChannelCount, + sourceFormat, sinkFormat, + sourceSampleRate, sinkSampleRate, + sourceStream->getSampleRateConversionQuality()); + + int32_t framesPerCallback = (sourceStream->getFramesPerCallback() == kUnspecified) + ? sourceStream->getFramesPerBurst() + : sourceStream->getFramesPerCallback(); + // Source + if ((sourceStream->getCallback() != nullptr && isOutput) + || (sourceStream->getCallback() == nullptr && !isOutput)) { + switch (sourceFormat) { + case AudioFormat::Float: + mSourceCaller = std::make_unique(sourceChannelCount, + framesPerCallback); + break; + case AudioFormat::I16: + mSourceCaller = std::make_unique(sourceChannelCount, + framesPerCallback); + break; + default: + LOGE("%s() Unsupported source caller format = %d", __func__, sourceFormat); + return Result::ErrorIllegalArgument; + } + mSourceCaller->setStream(sourceStream); + lastOutput = &mSourceCaller->output; + } else { + switch (sourceFormat) { + case AudioFormat::Float: + mSource = std::make_unique(sourceChannelCount); + break; + case AudioFormat::I16: + mSource = std::make_unique(sourceChannelCount); + break; + default: + LOGE("%s() Unsupported source format = %d", __func__, sourceFormat); + return Result::ErrorIllegalArgument; + } + if (!isOutput) { + // The BlockWriter is after the Sink so use the SinkStream size. + mBlockWriter.open(framesPerCallback * sinkStream->getBytesPerFrame()); + mAppBuffer = std::make_unique( + kDefaultBufferSize * sinkStream->getBytesPerFrame()); + } + lastOutput = &mSource->output; + } + + // Sample Rate conversion + if (sourceSampleRate != sinkSampleRate) { + mResampler.reset(MultiChannelResampler::make(sourceChannelCount, + sourceSampleRate, + sinkSampleRate, + convertOboeSRQualityToMCR( + sourceStream->getSampleRateConversionQuality()))); + mRateConverter = std::make_unique(sourceChannelCount, + *mResampler.get()); + lastOutput->connect(&mRateConverter->input); + lastOutput = &mRateConverter->output; + } + + // Expand the number of channels if required. + if (sourceChannelCount == 1 && sinkChannelCount > 1) { + mChannelConverter = std::make_unique(sinkChannelCount); + lastOutput->connect(&mChannelConverter->input); + lastOutput = &mChannelConverter->output; + } else if (sourceChannelCount != sinkChannelCount) { + LOGW("%s() Channel reduction not supported.", __func__); + return Result::ErrorUnimplemented; // TODO + } + + // Sink + switch (sinkFormat) { + case AudioFormat::Float: + mSink = std::make_unique(sinkChannelCount); + break; + case AudioFormat::I16: + mSink = std::make_unique(sinkChannelCount); + break; + default: + LOGE("%s() Unsupported sink format = %d", __func__, sinkFormat); + return Result::ErrorIllegalArgument;; + } + lastOutput->connect(&mSink->input); + + mFramePosition = 0; + + return Result::OK; +} + +int32_t DataConversionFlowGraph::read(void *buffer, int32_t numFrames, int64_t timeoutNanos) { + if (mSourceCaller) { + mSourceCaller->setTimeoutNanos(timeoutNanos); + } + int32_t numRead = mSink->read(mFramePosition, buffer, numFrames); + mFramePosition += numRead; + return numRead; +} + +// This is similar to pushing data through the flowgraph. +int32_t DataConversionFlowGraph::write(void *inputBuffer, int32_t numFrames) { + // Put the data from the Source at the head of the flowgraph. + mSource->setData(inputBuffer, numFrames); + while (true) { + // Pull and read some data in app format into a small buffer. + int32_t framesRead = mSink->read(mFramePosition, mAppBuffer.get(), flowgraph::kDefaultBufferSize); + mFramePosition += framesRead; + if (framesRead <= 0) break; + // Write to a block adapter, which will call the app whenever it has enough data. + int32_t bytesRead = mBlockWriter.write(mAppBuffer.get(), + framesRead * mFilterStream->getBytesPerFrame()); + if (bytesRead < 0) return bytesRead; // TODO review + } + return numFrames; +} + +int32_t DataConversionFlowGraph::onProcessFixedBlock(uint8_t *buffer, int32_t numBytes) { + int32_t numFrames = numBytes / mFilterStream->getBytesPerFrame(); + mCallbackResult = mFilterStream->getCallback()->onAudioReady(mFilterStream, buffer, numFrames); + // TODO handle STOP from callback, process data remaining in the block adapter + return numBytes; +} \ No newline at end of file diff --git a/modules/juce_audio_devices/native/oboe/src/common/DataConversionFlowGraph.h b/modules/juce_audio_devices/native/oboe/src/common/DataConversionFlowGraph.h new file mode 100644 index 0000000000..b1d5ebe8bc --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/common/DataConversionFlowGraph.h @@ -0,0 +1,84 @@ +/* + * Copyright (C) 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_OBOE_FLOW_GRAPH_H +#define OBOE_OBOE_FLOW_GRAPH_H + +#include +#include +#include + +#include +#include +#include +#include "AudioSourceCaller.h" +#include "FixedBlockWriter.h" + +namespace oboe { + +class AudioStream; +class AudioSourceCaller; + +/** + * Convert PCM channels, format and sample rate for optimal latency. + */ +class DataConversionFlowGraph : public FixedBlockProcessor { +public: + + DataConversionFlowGraph() + : mBlockWriter(*this) {} + + void setSource(const void *buffer, int32_t numFrames); + + /** Connect several modules together to convert from source to sink. + * This should only be called once for each instance. + * + * @param sourceFormat + * @param sourceChannelCount + * @param sinkFormat + * @param sinkChannelCount + * @return + */ + oboe::Result configure(oboe::AudioStream *sourceStream, oboe::AudioStream *sinkStream); + + int32_t read(void *buffer, int32_t numFrames, int64_t timeoutNanos); + + int32_t write(void *buffer, int32_t numFrames); + + int32_t onProcessFixedBlock(uint8_t *buffer, int32_t numBytes) override; + + DataCallbackResult getDataCallbackResult() { + return mCallbackResult; + } + +private: + std::unique_ptr mSource; + std::unique_ptr mSourceCaller; + std::unique_ptr mChannelConverter; + std::unique_ptr mResampler; + std::unique_ptr mRateConverter; + std::unique_ptr mSink; + + FixedBlockWriter mBlockWriter; + DataCallbackResult mCallbackResult = DataCallbackResult::Continue; + AudioStream *mFilterStream = nullptr; + std::unique_ptr mAppBuffer; + + int64_t mFramePosition = 0; +}; + +} +#endif //OBOE_OBOE_FLOW_GRAPH_H diff --git a/modules/juce_audio_devices/native/oboe/src/common/FilterAudioStream.cpp b/modules/juce_audio_devices/native/oboe/src/common/FilterAudioStream.cpp new file mode 100644 index 0000000000..2ca749bebd --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/common/FilterAudioStream.cpp @@ -0,0 +1,72 @@ +/* + * Copyright 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include + +#include "FilterAudioStream.h" + +using namespace oboe; +using namespace flowgraph; + +Result FilterAudioStream::configureFlowGraph() { + mFlowGraph = std::make_unique(); + bool isOutput = getDirection() == Direction::Output; + + AudioStream *sourceStream = isOutput ? this : mChildStream.get(); + AudioStream *sinkStream = isOutput ? mChildStream.get() : this; + + mRateScaler = ((double) sourceStream->getSampleRate()) / sinkStream->getSampleRate(); + + return mFlowGraph->configure(sourceStream, sinkStream); +} + +// Put the data to be written at the source end of the flowgraph. +// Then read (pull) the data from the flowgraph and write it to the +// child stream. +ResultWithValue FilterAudioStream::write(const void *buffer, + int32_t numFrames, + int64_t timeoutNanoseconds) { + int32_t framesWritten = 0; + mFlowGraph->setSource(buffer, numFrames); + while (true) { + int32_t numRead = mFlowGraph->read(mBlockingBuffer.get(), + getFramesPerBurst(), + timeoutNanoseconds); + if (numRead < 0) { + return ResultWithValue::createBasedOnSign(numRead); + } + if (numRead == 0) { + break; // finished processing the source buffer + } + auto writeResult = mChildStream->write(mBlockingBuffer.get(), + numRead, + timeoutNanoseconds); + if (!writeResult) { + return writeResult; + } + framesWritten += writeResult.value(); + } + return ResultWithValue::createBasedOnSign(framesWritten); +} + +// Read (pull) the data we want from the sink end of the flowgraph. +// The necessary data will be read from the child stream using a flowgraph callback. +ResultWithValue FilterAudioStream::read(void *buffer, + int32_t numFrames, + int64_t timeoutNanoseconds) { + int32_t framesRead = mFlowGraph->read(buffer, numFrames, timeoutNanoseconds); + return ResultWithValue::createBasedOnSign(framesRead); +} diff --git a/modules/juce_audio_devices/native/oboe/src/common/FilterAudioStream.h b/modules/juce_audio_devices/native/oboe/src/common/FilterAudioStream.h new file mode 100644 index 0000000000..6baccd8d6a --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/common/FilterAudioStream.h @@ -0,0 +1,204 @@ +/* + * Copyright 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_FILTER_AUDIO_STREAM_H +#define OBOE_FILTER_AUDIO_STREAM_H + +#include +#include +#include "DataConversionFlowGraph.h" + +namespace oboe { + +/** + * An AudioStream that wraps another AudioStream and provides audio data conversion. + * Operations may include channel conversion, data format conversion and/or sample rate conversion. + */ +class FilterAudioStream : public AudioStream, AudioStreamCallback { +public: + + /** + * Construct an `AudioStream` using the given `AudioStreamBuilder` and a child AudioStream. + * + * This should only be called internally by AudioStreamBuilder. + * Ownership of childStream will be passed to this object. + * + * @param builder containing all the stream's attributes + */ + FilterAudioStream(const AudioStreamBuilder &builder, AudioStream *childStream) + : AudioStream(builder) + , mChildStream(childStream) { + // Intercept the callback if used. + if (builder.getCallback() != nullptr) { + mStreamCallback = mChildStream->swapCallback(this); + } else { + const int size = childStream->getFramesPerBurst() * childStream->getBytesPerFrame(); + mBlockingBuffer = std::make_unique(size); + } + + // Copy parameters that may not match builder. + mBufferCapacityInFrames = mChildStream->getBufferCapacityInFrames(); + mPerformanceMode = mChildStream->getPerformanceMode(); + } + + virtual ~FilterAudioStream() = default; + + AudioStream *getChildStream() const { + return mChildStream.get(); + } + + Result configureFlowGraph(); + + // Close child and parent. + Result close() override { + const Result result1 = mChildStream->close(); + const Result result2 = AudioStream::close(); + return (result1 != Result::OK ? result1 : result2); + } + + /** + * Start the stream asynchronously. Returns immediately (does not block). Equivalent to calling + * `start(0)`. + */ + Result requestStart() override { + return mChildStream->requestStart(); + } + + /** + * Pause the stream asynchronously. Returns immediately (does not block). Equivalent to calling + * `pause(0)`. + */ + Result requestPause() override { + return mChildStream->requestPause(); + } + + /** + * Flush the stream asynchronously. Returns immediately (does not block). Equivalent to calling + * `flush(0)`. + */ + Result requestFlush() override { + return mChildStream->requestFlush(); + } + + /** + * Stop the stream asynchronously. Returns immediately (does not block). Equivalent to calling + * `stop(0)`. + */ + Result requestStop() override { + return mChildStream->requestStop(); + } + + ResultWithValue read(void *buffer, + int32_t numFrames, + int64_t timeoutNanoseconds) override; + + ResultWithValue write(const void *buffer, + int32_t numFrames, + int64_t timeoutNanoseconds) override; + + StreamState getState() const override { + return mChildStream->getState(); + } + + Result waitForStateChange( + StreamState inputState, + StreamState *nextState, + int64_t timeoutNanoseconds) override { + return mChildStream->waitForStateChange(inputState, nextState, timeoutNanoseconds); + } + + bool isXRunCountSupported() const override { + return mChildStream->isXRunCountSupported(); + } + + int32_t getFramesPerBurst() override { + return mChildStream->getFramesPerBurst(); + } + + AudioApi getAudioApi() const override { + return mChildStream->getAudioApi(); + } + + void updateFramesWritten() override { + // TODO for output, just count local writes? + mFramesWritten = static_cast(mChildStream->getFramesWritten() * mRateScaler); + } + + void updateFramesRead() override { + // TODO for input, just count local reads? + mFramesRead = static_cast(mChildStream->getFramesRead() * mRateScaler); + } + + void *getUnderlyingStream() const override { + return mChildStream->getUnderlyingStream(); + } + + ResultWithValue setBufferSizeInFrames(int32_t requestedFrames) override { + return mChildStream->setBufferSizeInFrames(requestedFrames); + } + + int32_t getBufferSizeInFrames() override { + mBufferSizeInFrames = mChildStream->getBufferSizeInFrames(); + return mBufferSizeInFrames; + } + + ResultWithValue getXRunCount() const override { + return mChildStream->getXRunCount(); + } + + ResultWithValue calculateLatencyMillis() override { + // This will automatically include the latency of the flowgraph? + return mChildStream->calculateLatencyMillis(); + } + + Result getTimestamp(clockid_t clockId, + int64_t *framePosition, + int64_t *timeNanoseconds) override { + int64_t childPosition = 0; + Result result = mChildStream->getTimestamp(clockId, &childPosition, timeNanoseconds); + *framePosition = childPosition * mRateScaler; + return result; + } + + DataCallbackResult onAudioReady(AudioStream *oboeStream, + void *audioData, + int32_t numFrames) override { + int32_t framesProcessed; + if (oboeStream->getDirection() == Direction::Output) { + framesProcessed = mFlowGraph->read(audioData, numFrames, 0 /* timeout */); + } else { + framesProcessed = mFlowGraph->write(audioData, numFrames); + } + return (framesProcessed < numFrames) + ? DataCallbackResult::Stop + : mFlowGraph->getDataCallbackResult(); + } + + void onErrorBeforeClose(AudioStream *oboeStream, Result error) override {} + + void onErrorAfterClose(AudioStream *oboeStream, Result error) override {} + +private: + + std::unique_ptr mChildStream; // this stream wraps the child stream + std::unique_ptr mFlowGraph; // for converting data + std::unique_ptr mBlockingBuffer; // temp buffer for write() + double mRateScaler = 1.0; // ratio parent/child sample rates +}; + +} // oboe + +#endif //OBOE_FILTER_AUDIO_STREAM_H diff --git a/modules/juce_audio_devices/native/oboe/src/common/FixedBlockAdapter.cpp b/modules/juce_audio_devices/native/oboe/src/common/FixedBlockAdapter.cpp new file mode 100644 index 0000000000..e14c2caadd --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/common/FixedBlockAdapter.cpp @@ -0,0 +1,38 @@ +/* + * Copyright (C) 2017 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include + +#include "FixedBlockAdapter.h" + +FixedBlockAdapter::~FixedBlockAdapter() { +} + +int32_t FixedBlockAdapter::open(int32_t bytesPerFixedBlock) +{ + mSize = bytesPerFixedBlock; + mStorage = std::make_unique(bytesPerFixedBlock); + mPosition = 0; + return 0; +} + +int32_t FixedBlockAdapter::close() +{ + mStorage.reset(nullptr); + mSize = 0; + mPosition = 0; + return 0; +} diff --git a/modules/juce_audio_devices/native/oboe/src/common/FixedBlockAdapter.h b/modules/juce_audio_devices/native/oboe/src/common/FixedBlockAdapter.h new file mode 100644 index 0000000000..99a3ee0e46 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/common/FixedBlockAdapter.h @@ -0,0 +1,66 @@ +/* + * Copyright (C) 2017 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef AAUDIO_FIXED_BLOCK_ADAPTER_H +#define AAUDIO_FIXED_BLOCK_ADAPTER_H + +#include +#include + +/** + * Interface for a class that needs fixed-size blocks. + */ +class FixedBlockProcessor { +public: + virtual ~FixedBlockProcessor() = default; + /** + * + * @param buffer Pointer to first byte of data. + * @param numBytes This will be a fixed size specified in FixedBlockAdapter::open(). + * @return Number of bytes processed or a negative error code. + */ + virtual int32_t onProcessFixedBlock(uint8_t *buffer, int32_t numBytes) = 0; +}; + +/** + * Base class for a variable-to-fixed-size block adapter. + */ +class FixedBlockAdapter +{ +public: + FixedBlockAdapter(FixedBlockProcessor &fixedBlockProcessor) + : mFixedBlockProcessor(fixedBlockProcessor) {} + + virtual ~FixedBlockAdapter(); + + /** + * Allocate internal resources needed for buffering data. + */ + virtual int32_t open(int32_t bytesPerFixedBlock); + + /** + * Free internal resources. + */ + int32_t close(); + +protected: + FixedBlockProcessor &mFixedBlockProcessor; + std::unique_ptr mStorage; // Store data here while assembling buffers. + int32_t mSize = 0; // Size in bytes of the fixed size buffer. + int32_t mPosition = 0; // Offset of the last byte read or written. +}; + +#endif /* AAUDIO_FIXED_BLOCK_ADAPTER_H */ diff --git a/modules/juce_audio_devices/native/oboe/src/common/FixedBlockReader.cpp b/modules/juce_audio_devices/native/oboe/src/common/FixedBlockReader.cpp new file mode 100644 index 0000000000..62e9664d9f --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/common/FixedBlockReader.cpp @@ -0,0 +1,73 @@ +/* + * Copyright (C) 2017 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include +#include + +#include "FixedBlockAdapter.h" + +#include "FixedBlockReader.h" + + +FixedBlockReader::FixedBlockReader(FixedBlockProcessor &fixedBlockProcessor) + : FixedBlockAdapter(fixedBlockProcessor) { + mPosition = mSize; +} + +int32_t FixedBlockReader::open(int32_t bytesPerFixedBlock) { + int32_t result = FixedBlockAdapter::open(bytesPerFixedBlock); + mPosition = 0; + mValid = 0; + return result; +} + +int32_t FixedBlockReader::readFromStorage(uint8_t *buffer, int32_t numBytes) { + int32_t bytesToRead = numBytes; + int32_t dataAvailable = mValid - mPosition; + if (bytesToRead > dataAvailable) { + bytesToRead = dataAvailable; + } + memcpy(buffer, mStorage.get() + mPosition, bytesToRead); + mPosition += bytesToRead; + return bytesToRead; +} + +int32_t FixedBlockReader::read(uint8_t *buffer, int32_t numBytes) { + int32_t bytesRead; + int32_t bytesLeft = numBytes; + while(bytesLeft > 0) { + if (mPosition < mValid) { + // Use up bytes currently in storage. + bytesRead = readFromStorage(buffer, bytesLeft); + buffer += bytesRead; + bytesLeft -= bytesRead; + } else if (bytesLeft >= mSize) { + // Nothing in storage. Read through if enough for a complete block. + bytesRead = mFixedBlockProcessor.onProcessFixedBlock(buffer, mSize); + if (bytesRead < 0) return bytesRead; + buffer += bytesRead; + bytesLeft -= bytesRead; + } else { + // Just need a partial block so we have to reload storage. + bytesRead = mFixedBlockProcessor.onProcessFixedBlock(mStorage.get(), mSize); + if (bytesRead < 0) return bytesRead; + mPosition = 0; + mValid = bytesRead; + if (bytesRead == 0) break; + } + } + return numBytes - bytesLeft; +} diff --git a/modules/juce_audio_devices/native/oboe/src/common/FixedBlockReader.h b/modules/juce_audio_devices/native/oboe/src/common/FixedBlockReader.h new file mode 100644 index 0000000000..4cea9c8b4a --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/common/FixedBlockReader.h @@ -0,0 +1,60 @@ +/* + * Copyright (C) 2017 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef AAUDIO_FIXED_BLOCK_READER_H +#define AAUDIO_FIXED_BLOCK_READER_H + +#include + +#include "FixedBlockAdapter.h" + +/** + * Read from a fixed-size block to a variable sized block. + * + * This can be used to convert a pull data flow from fixed sized buffers to variable sized buffers. + * An example would be an audio output callback that reads from the app. + */ +class FixedBlockReader : public FixedBlockAdapter +{ +public: + FixedBlockReader(FixedBlockProcessor &fixedBlockProcessor); + + virtual ~FixedBlockReader() = default; + + int32_t open(int32_t bytesPerFixedBlock) override; + + /** + * Read into a variable sized block. + * + * Note that if the fixed-sized blocks must be aligned, then the variable-sized blocks + * must have the same alignment. + * For example, if the fixed-size blocks must be a multiple of 8, then the variable-sized + * blocks must also be a multiple of 8. + * + * @param buffer + * @param numBytes + * @return Number of bytes read or a negative error code. + */ + int32_t read(uint8_t *buffer, int32_t numBytes); + +private: + int32_t readFromStorage(uint8_t *buffer, int32_t numBytes); + + int32_t mValid = 0; // Number of valid bytes in mStorage. +}; + + +#endif /* AAUDIO_FIXED_BLOCK_READER_H */ diff --git a/modules/juce_audio_devices/native/oboe/src/common/FixedBlockWriter.cpp b/modules/juce_audio_devices/native/oboe/src/common/FixedBlockWriter.cpp new file mode 100644 index 0000000000..b3156097c8 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/common/FixedBlockWriter.cpp @@ -0,0 +1,73 @@ +/* + * Copyright (C) 2017 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include +#include + +#include "FixedBlockAdapter.h" +#include "FixedBlockWriter.h" + +FixedBlockWriter::FixedBlockWriter(FixedBlockProcessor &fixedBlockProcessor) + : FixedBlockAdapter(fixedBlockProcessor) {} + + +int32_t FixedBlockWriter::writeToStorage(uint8_t *buffer, int32_t numBytes) { + int32_t bytesToStore = numBytes; + int32_t roomAvailable = mSize - mPosition; + if (bytesToStore > roomAvailable) { + bytesToStore = roomAvailable; + } + memcpy(mStorage.get() + mPosition, buffer, bytesToStore); + mPosition += bytesToStore; + return bytesToStore; +} + +int32_t FixedBlockWriter::write(uint8_t *buffer, int32_t numBytes) { + int32_t bytesLeft = numBytes; + + // If we already have data in storage then add to it. + if (mPosition > 0) { + int32_t bytesWritten = writeToStorage(buffer, bytesLeft); + buffer += bytesWritten; + bytesLeft -= bytesWritten; + // If storage full then flush it out + if (mPosition == mSize) { + bytesWritten = mFixedBlockProcessor.onProcessFixedBlock(mStorage.get(), mSize); + if (bytesWritten < 0) return bytesWritten; + mPosition = 0; + if (bytesWritten < mSize) { + // Only some of the data was written! This should not happen. + return -1; + } + } + } + + // Write through if enough for a complete block. + while(bytesLeft > mSize) { + int32_t bytesWritten = mFixedBlockProcessor.onProcessFixedBlock(buffer, mSize); + if (bytesWritten < 0) return bytesWritten; + buffer += bytesWritten; + bytesLeft -= bytesWritten; + } + + // Save any remaining partial blocks for next time. + if (bytesLeft > 0) { + int32_t bytesWritten = writeToStorage(buffer, bytesLeft); + bytesLeft -= bytesWritten; + } + + return numBytes - bytesLeft; +} diff --git a/modules/juce_audio_devices/native/oboe/src/common/FixedBlockWriter.h b/modules/juce_audio_devices/native/oboe/src/common/FixedBlockWriter.h new file mode 100644 index 0000000000..6dea6ca96d --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/common/FixedBlockWriter.h @@ -0,0 +1,54 @@ +/* + * Copyright (C) 2017 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef AAUDIO_FIXED_BLOCK_WRITER_H +#define AAUDIO_FIXED_BLOCK_WRITER_H + +#include + +#include "FixedBlockAdapter.h" + +/** + * This can be used to convert a push data flow from variable sized buffers to fixed sized buffers. + * An example would be an audio input callback. + */ +class FixedBlockWriter : public FixedBlockAdapter +{ +public: + FixedBlockWriter(FixedBlockProcessor &fixedBlockProcessor); + + virtual ~FixedBlockWriter() = default; + + /** + * Write from a variable sized block. + * + * Note that if the fixed-sized blocks must be aligned, then the variable-sized blocks + * must have the same alignment. + * For example, if the fixed-size blocks must be a multiple of 8, then the variable-sized + * blocks must also be a multiple of 8. + * + * @param buffer + * @param numBytes + * @return Number of bytes written or a negative error code. + */ + int32_t write(uint8_t *buffer, int32_t numBytes); + +private: + + int32_t writeToStorage(uint8_t *buffer, int32_t numBytes); +}; + +#endif /* AAUDIO_FIXED_BLOCK_WRITER_H */ diff --git a/modules/juce_audio_devices/native/oboe/src/common/LatencyTuner.cpp b/modules/juce_audio_devices/native/oboe/src/common/LatencyTuner.cpp new file mode 100644 index 0000000000..cce8e18976 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/common/LatencyTuner.cpp @@ -0,0 +1,102 @@ +/* + * Copyright 2017 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include "oboe/LatencyTuner.h" + +using namespace oboe; + +LatencyTuner::LatencyTuner(AudioStream &stream) + : LatencyTuner(stream, stream.getBufferCapacityInFrames()){ + } + +LatencyTuner::LatencyTuner(oboe::AudioStream &stream, int32_t maximumBufferSize) + : mStream(stream) + , mMaxBufferSize(maximumBufferSize) { + reset(); +} + +Result LatencyTuner::tune() { + if (mState == State::Unsupported) { + return Result::ErrorUnimplemented; + } + + Result result = Result::OK; + + // Process reset requests. + int32_t numRequests = mLatencyTriggerRequests.load(); + if (numRequests != mLatencyTriggerResponses.load()) { + mLatencyTriggerResponses.store(numRequests); + reset(); + } + + // Set state to Active if the idle countdown has reached zero. + if (mState == State::Idle && --mIdleCountDown <= 0) { + mState = State::Active; + } + + // When state is Active attempt to change the buffer size if the number of xRuns has increased. + if (mState == State::Active) { + + auto xRunCountResult = mStream.getXRunCount(); + if (xRunCountResult == Result::OK) { + if ((xRunCountResult.value() - mPreviousXRuns) > 0) { + mPreviousXRuns = xRunCountResult.value(); + int32_t oldBufferSize = mStream.getBufferSizeInFrames(); + int32_t requestedBufferSize = oldBufferSize + mStream.getFramesPerBurst(); + + // Do not request more than the maximum buffer size (which was either user-specified + // or was from stream->getBufferCapacityInFrames()) + if (requestedBufferSize > mMaxBufferSize) requestedBufferSize = mMaxBufferSize; + + auto setBufferResult = mStream.setBufferSizeInFrames(requestedBufferSize); + if (setBufferResult != Result::OK) { + result = setBufferResult; + mState = State::Unsupported; + } else if (setBufferResult.value() == oldBufferSize) { + mState = State::AtMax; + } + } + } else { + mState = State::Unsupported; + } + } + + if (mState == State::Unsupported) { + result = Result::ErrorUnimplemented; + } + + if (mState == State::AtMax) { + result = Result::OK; + } + return result; +} + +void LatencyTuner::requestReset() { + if (mState != State::Unsupported) { + mLatencyTriggerRequests++; + } +} + +void LatencyTuner::reset() { + mState = State::Idle; + mIdleCountDown = kIdleCount; + // Set to minimal latency + mStream.setBufferSizeInFrames(2 * mStream.getFramesPerBurst()); +} + +bool LatencyTuner::isAtMaximumBufferSize() { + return mState == State::AtMax; +} diff --git a/modules/juce_audio_devices/native/oboe/src/common/MonotonicCounter.h b/modules/juce_audio_devices/native/oboe/src/common/MonotonicCounter.h new file mode 100644 index 0000000000..00c979c50e --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/common/MonotonicCounter.h @@ -0,0 +1,112 @@ +/* + * Copyright 2016 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef COMMON_MONOTONIC_COUNTER_H +#define COMMON_MONOTONIC_COUNTER_H + +#include + +/** + * Maintain a 64-bit monotonic counter. + * Can be used to track a 32-bit counter that wraps or gets reset. + * + * Note that this is not atomic and has no interior locks. + * A caller will need to provide their own exterior locking + * if they need to use it from multiple threads. + */ +class MonotonicCounter { + +public: + MonotonicCounter() {} + virtual ~MonotonicCounter() {} + + /** + * @return current value of the counter + */ + int64_t get() const { + return mCounter64; + } + + /** + * set the current value of the counter + */ + void set(int64_t counter) { + mCounter64 = counter; + } + + /** + * Advance the counter if delta is positive. + * @return current value of the counter + */ + int64_t increment(int64_t delta) { + if (delta > 0) { + mCounter64 += delta; + } + return mCounter64; + } + + /** + * Advance the 64-bit counter if (current32 - previousCurrent32) > 0. + * This can be used to convert a 32-bit counter that may be wrapping into + * a monotonic 64-bit counter. + * + * This counter32 should NOT be allowed to advance by more than 0x7FFFFFFF between calls. + * Think of the wrapping counter like a sine wave. If the frequency of the signal + * is more than half the sampling rate (Nyquist rate) then you cannot measure it properly. + * If the counter wraps around every 24 hours then we should measure it with a period + * of less than 12 hours. + * + * @return current value of the 64-bit counter + */ + int64_t update32(int32_t counter32) { + int32_t delta = counter32 - mCounter32; + // protect against the mCounter64 going backwards + if (delta > 0) { + mCounter64 += delta; + mCounter32 = counter32; + } + return mCounter64; + } + + /** + * Reset the stored value of the 32-bit counter. + * This is used if your counter32 has been reset to zero. + */ + void reset32() { + mCounter32 = 0; + } + + /** + * Round 64-bit counter up to a multiple of the period. + * + * The period must be positive. + * + * @param period might be, for example, a buffer capacity + */ + void roundUp64(int32_t period) { + if (period > 0) { + int64_t numPeriods = (mCounter64 + period - 1) / period; + mCounter64 = numPeriods * period; + } + } + +private: + int64_t mCounter64 = 0; + int32_t mCounter32 = 0; +}; + + +#endif //COMMON_MONOTONIC_COUNTER_H diff --git a/modules/juce_audio_devices/native/oboe/src/common/OboeDebug.h b/modules/juce_audio_devices/native/oboe/src/common/OboeDebug.h new file mode 100644 index 0000000000..e5ba9e3b0f --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/common/OboeDebug.h @@ -0,0 +1,46 @@ +/* + * Copyright (C) 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + * + */ + +#ifndef OBOE_DEBUG_H +#define OBOE_DEBUG_H + +#if OBOE_ENABLE_LOGGING + +#include + +#ifndef MODULE_NAME +#define MODULE_NAME "OboeAudio" +#endif + +#define LOGV(...) __android_log_print(ANDROID_LOG_VERBOSE, MODULE_NAME, __VA_ARGS__) +#define LOGD(...) __android_log_print(ANDROID_LOG_DEBUG, MODULE_NAME, __VA_ARGS__) +#define LOGI(...) __android_log_print(ANDROID_LOG_INFO, MODULE_NAME, __VA_ARGS__) +#define LOGW(...) __android_log_print(ANDROID_LOG_WARN, MODULE_NAME, __VA_ARGS__) +#define LOGE(...) __android_log_print(ANDROID_LOG_ERROR, MODULE_NAME, __VA_ARGS__) +#define LOGF(...) __android_log_print(ANDROID_LOG_FATAL, MODULE_NAME, __VA_ARGS__) + +#else + +#define LOGV(...) +#define LOGD(...) +#define LOGI(...) +#define LOGW(...) +#define LOGE(...) +#define LOGF(...) +#endif + +#endif //OBOE_DEBUG_H diff --git a/modules/juce_audio_devices/native/oboe/src/common/QuirksManager.cpp b/modules/juce_audio_devices/native/oboe/src/common/QuirksManager.cpp new file mode 100644 index 0000000000..2d5f644f7d --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/common/QuirksManager.cpp @@ -0,0 +1,70 @@ +/* + * Copyright 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include +#include +#include "QuirksManager.h" + +using namespace oboe; + +bool QuirksManager::isConversionNeeded( + const AudioStreamBuilder &builder, + AudioStreamBuilder &childBuilder) { + bool conversionNeeded = false; + const bool isLowLatency = builder.getPerformanceMode() == PerformanceMode::LowLatency; + const bool isInput = builder.getDirection() == Direction::Input; + const bool isFloat = builder.getFormat() == AudioFormat::Float; + + // If a SAMPLE RATE is specified for low latency then let the native code choose an optimal rate. + // TODO There may be a problem if the devices supports low latency + // at a higher rate than the default. + if (builder.getSampleRate() != oboe::Unspecified + && builder.getSampleRateConversionQuality() != SampleRateConversionQuality::None + && isLowLatency + ) { + childBuilder.setSampleRate(oboe::Unspecified); // native API decides the best sample rate + conversionNeeded = true; + } + + // Data Format + // OpenSL ES and AAudio before P do not support FAST path for FLOAT capture. + if (isFloat + && isInput + && builder.isFormatConversionAllowed() + && isLowLatency + && (!builder.willUseAAudio() || (getSdkVersion() < __ANDROID_API_P__)) + ) { + childBuilder.setFormat(AudioFormat::I16); // needed for FAST track + conversionNeeded = true; + } + + // Channel Count + if (builder.getChannelCount() != oboe::Unspecified + && builder.isChannelConversionAllowed()) { + if (builder.getChannelCount() == 2 // stereo? + && isInput + && isLowLatency + && (!builder.willUseAAudio() && (getSdkVersion() == __ANDROID_API_O__))) { + // workaround for temporary heap size regression, b/66967812 + childBuilder.setChannelCount(1); + conversionNeeded = true; + } + // Note that MMAP does not support mono in 8.1. But that would only matter on Pixel 1 + // phones and they have almost all been updated to 9.0. + } + + return conversionNeeded; +} diff --git a/modules/juce_audio_devices/native/oboe/src/common/QuirksManager.h b/modules/juce_audio_devices/native/oboe/src/common/QuirksManager.h new file mode 100644 index 0000000000..d14f1d3d4c --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/common/QuirksManager.h @@ -0,0 +1,52 @@ +/* + * Copyright 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_QUIRKS_MANAGER_H +#define OBOE_QUIRKS_MANAGER_H + +#include +#include + +namespace oboe { + +/** + * Based on manufacturer, model and Android version number + * decide whether data conversion needs to occur. + * + * This also manages device and version specific workarounds. + */ + +class QuirksManager { +public: + + static QuirksManager &getInstance() { + static QuirksManager instance; + return instance; + } + + /** + * + * @param builder builder provided by application + * @param childBuilder modified builder appropriate for the underlying device + * @return true if conversion is needed + */ + bool isConversionNeeded(const AudioStreamBuilder &builder, AudioStreamBuilder &childBuilder); + +private: +}; + +} +#endif //OBOE_QUIRKS_MANAGER_H diff --git a/modules/juce_audio_devices/native/oboe/src/common/SourceFloatCaller.cpp b/modules/juce_audio_devices/native/oboe/src/common/SourceFloatCaller.cpp new file mode 100644 index 0000000000..631af85342 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/common/SourceFloatCaller.cpp @@ -0,0 +1,30 @@ +/* + * Copyright 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include +#include +#include "flowgraph/FlowGraphNode.h" +#include "SourceFloatCaller.h" + +using namespace oboe; +using namespace flowgraph; + +int32_t SourceFloatCaller::onProcess(int32_t numFrames) { + int32_t numBytes = mStream->getBytesPerFrame() * numFrames; + int32_t bytesRead = mBlockReader.read((uint8_t *) output.getBuffer(), numBytes); + int32_t framesRead = bytesRead / mStream->getBytesPerFrame(); + return framesRead; +} diff --git a/modules/juce_audio_devices/native/oboe/src/common/SourceFloatCaller.h b/modules/juce_audio_devices/native/oboe/src/common/SourceFloatCaller.h new file mode 100644 index 0000000000..85a158550a --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/common/SourceFloatCaller.h @@ -0,0 +1,44 @@ +/* + * Copyright 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_SOURCE_FLOAT_CALLER_H +#define OBOE_SOURCE_FLOAT_CALLER_H + +#include +#include + +#include "flowgraph/FlowGraphNode.h" +#include "AudioSourceCaller.h" +#include "FixedBlockReader.h" + +namespace oboe { +/** + * AudioSource that uses callback to get more float data. + */ +class SourceFloatCaller : public AudioSourceCaller { +public: + SourceFloatCaller(int32_t channelCount, int32_t framesPerCallback) + : AudioSourceCaller(channelCount, framesPerCallback, (int32_t)sizeof(float)) {} + + int32_t onProcess(int32_t numFrames) override; + + const char *getName() override { + return "SourceFloatCaller"; + } +}; + +} +#endif //OBOE_SOURCE_FLOAT_CALLER_H diff --git a/modules/juce_audio_devices/native/oboe/src/common/SourceI16Caller.cpp b/modules/juce_audio_devices/native/oboe/src/common/SourceI16Caller.cpp new file mode 100644 index 0000000000..2ab372b638 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/common/SourceI16Caller.cpp @@ -0,0 +1,47 @@ +/* + * Copyright 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include +#include +#include "flowgraph/FlowGraphNode.h" +#include "SourceI16Caller.h" + +#if FLOWGRAPH_ANDROID_INTERNAL +#include +#endif + +using namespace oboe; +using namespace flowgraph; + +int32_t SourceI16Caller::onProcess(int32_t numFrames) { + int32_t numBytes = mStream->getBytesPerFrame() * numFrames; + int32_t bytesRead = mBlockReader.read((uint8_t *) mConversionBuffer.get(), numBytes); + int32_t framesRead = bytesRead / mStream->getBytesPerFrame(); + + float *floatData = output.getBuffer(); + const int16_t *shortData = mConversionBuffer.get(); + int32_t numSamples = framesRead * output.getSamplesPerFrame(); + +#if FLOWGRAPH_ANDROID_INTERNAL + memcpy_to_float_from_i16(floatData, shortData, numSamples); +#else + for (int i = 0; i < numSamples; i++) { + *floatData++ = *shortData++ * (1.0f / 32768); + } +#endif + + return framesRead; +} diff --git a/modules/juce_audio_devices/native/oboe/src/common/SourceI16Caller.h b/modules/juce_audio_devices/native/oboe/src/common/SourceI16Caller.h new file mode 100644 index 0000000000..22c1b9aba3 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/common/SourceI16Caller.h @@ -0,0 +1,48 @@ +/* + * Copyright 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_SOURCE_I16_CALLER_H +#define OBOE_SOURCE_I16_CALLER_H + +#include +#include + +#include "flowgraph/FlowGraphNode.h" +#include "AudioSourceCaller.h" +#include "FixedBlockReader.h" + +namespace oboe { +/** + * AudioSource that uses callback to get more data. + */ +class SourceI16Caller : public AudioSourceCaller { +public: + SourceI16Caller(int32_t channelCount, int32_t framesPerCallback) + : AudioSourceCaller(channelCount, framesPerCallback, sizeof(int16_t)) { + mConversionBuffer = std::make_unique(channelCount * output.getFramesPerBuffer()); + } + + int32_t onProcess(int32_t numFrames) override; + + const char *getName() override { + return "SourceI16Caller"; + } +private: + std::unique_ptr mConversionBuffer; +}; + +} +#endif //OBOE_SOURCE_I16_CALLER_H diff --git a/modules/juce_audio_devices/native/oboe/src/common/StabilizedCallback.cpp b/modules/juce_audio_devices/native/oboe/src/common/StabilizedCallback.cpp new file mode 100644 index 0000000000..692db89ba6 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/common/StabilizedCallback.cpp @@ -0,0 +1,112 @@ +/* + * Copyright (C) 2018 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include "oboe/StabilizedCallback.h" +#include "common/AudioClock.h" +#include "common/Trace.h" + +constexpr int32_t kLoadGenerationStepSizeNanos = 20000; +constexpr float kPercentageOfCallbackToUse = 0.8; + +using namespace oboe; + +StabilizedCallback::StabilizedCallback(AudioStreamCallback *callback) : mCallback(callback){ + Trace::initialize(); +} + +/** + * An audio callback which attempts to do work for a fixed amount of time. + * + * @param oboeStream + * @param audioData + * @param numFrames + * @return + */ +DataCallbackResult +StabilizedCallback::onAudioReady(AudioStream *oboeStream, void *audioData, int32_t numFrames) { + + int64_t startTimeNanos = AudioClock::getNanoseconds(); + + if (mFrameCount == 0){ + mEpochTimeNanos = startTimeNanos; + } + + int64_t durationSinceEpochNanos = startTimeNanos - mEpochTimeNanos; + + // In an ideal world the callback start time will be exactly the same as the duration of the + // frames already read/written into the stream. In reality the callback can start early + // or late. By finding the delta we can calculate the target duration for our stabilized + // callback. + int64_t idealStartTimeNanos = (mFrameCount * kNanosPerSecond) / oboeStream->getSampleRate(); + int64_t lateStartNanos = durationSinceEpochNanos - idealStartTimeNanos; + + if (lateStartNanos < 0){ + // This was an early start which indicates that our previous epoch was a late callback. + // Update our epoch to this more accurate time. + mEpochTimeNanos = startTimeNanos; + mFrameCount = 0; + } + + int64_t numFramesAsNanos = (numFrames * kNanosPerSecond) / oboeStream->getSampleRate(); + int64_t targetDurationNanos = static_cast( + (numFramesAsNanos * kPercentageOfCallbackToUse) - lateStartNanos); + + Trace::beginSection("Actual load"); + DataCallbackResult result = mCallback->onAudioReady(oboeStream, audioData, numFrames); + Trace::endSection(); + + int64_t executionDurationNanos = AudioClock::getNanoseconds() - startTimeNanos; + int64_t stabilizingLoadDurationNanos = targetDurationNanos - executionDurationNanos; + + Trace::beginSection("Stabilized load for %lldns", stabilizingLoadDurationNanos); + generateLoad(stabilizingLoadDurationNanos); + Trace::endSection(); + + // Wraparound: At 48000 frames per second mFrameCount wraparound will occur after 6m years, + // significantly longer than the average lifetime of an Android phone. + mFrameCount += numFrames; + return result; +} + +void StabilizedCallback::generateLoad(int64_t durationNanos) { + + int64_t currentTimeNanos = AudioClock::getNanoseconds(); + int64_t deadlineTimeNanos = currentTimeNanos + durationNanos; + + // opsPerStep gives us an estimated number of operations which need to be run to fully utilize + // the CPU for a fixed amount of time (specified by kLoadGenerationStepSizeNanos). + // After each step the opsPerStep value is re-calculated based on the actual time taken to + // execute those operations. + auto opsPerStep = (int)(mOpsPerNano * kLoadGenerationStepSizeNanos); + int64_t stepDurationNanos = 0; + int64_t previousTimeNanos = 0; + + while (currentTimeNanos <= deadlineTimeNanos){ + + for (int i = 0; i < opsPerStep; i++) cpu_relax(); + + previousTimeNanos = currentTimeNanos; + currentTimeNanos = AudioClock::getNanoseconds(); + stepDurationNanos = currentTimeNanos - previousTimeNanos; + + // Calculate exponential moving average to smooth out values, this acts as a low pass filter. + // @see https://en.wikipedia.org/wiki/Moving_average#Exponential_moving_average + static const float kFilterCoefficient = 0.1; + auto measuredOpsPerNano = (double) opsPerStep / stepDurationNanos; + mOpsPerNano = kFilterCoefficient * measuredOpsPerNano + (1.0 - kFilterCoefficient) * mOpsPerNano; + opsPerStep = (int) (mOpsPerNano * kLoadGenerationStepSizeNanos); + } +} \ No newline at end of file diff --git a/modules/juce_audio_devices/native/oboe/src/common/Trace.cpp b/modules/juce_audio_devices/native/oboe/src/common/Trace.cpp new file mode 100644 index 0000000000..5ed445b5dc --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/common/Trace.cpp @@ -0,0 +1,75 @@ +/* + * Copyright 2018 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include +#include +#include "Trace.h" +#include "OboeDebug.h" + +static char buffer[256]; + +// Tracing functions +static void *(*ATrace_beginSection)(const char *sectionName); + +static void *(*ATrace_endSection)(); + +typedef void *(*fp_ATrace_beginSection)(const char *sectionName); + +typedef void *(*fp_ATrace_endSection)(); + +bool Trace::mIsTracingSupported = false; + +void Trace::beginSection(const char *format, ...){ + + if (mIsTracingSupported) { + va_list va; + va_start(va, format); + vsprintf(buffer, format, va); + ATrace_beginSection(buffer); + va_end(va); + } else { + LOGE("Tracing is either not initialized (call Trace::initialize()) " + "or not supported on this device"); + } +} + +void Trace::endSection() { + + if (mIsTracingSupported) { + ATrace_endSection(); + } +} + +void Trace::initialize() { + + // Using dlsym allows us to use tracing on API 21+ without needing android/trace.h which wasn't + // published until API 23 + void *lib = dlopen("libandroid.so", RTLD_NOW | RTLD_LOCAL); + if (lib == nullptr) { + LOGE("Could not open libandroid.so to dynamically load tracing symbols"); + } else { + ATrace_beginSection = + reinterpret_cast( + dlsym(lib, "ATrace_beginSection")); + ATrace_endSection = + reinterpret_cast( + dlsym(lib, "ATrace_endSection")); + + if (ATrace_beginSection != nullptr && ATrace_endSection != nullptr){ + mIsTracingSupported = true; + } + } +} \ No newline at end of file diff --git a/modules/juce_audio_devices/native/oboe/src/common/Trace.h b/modules/juce_audio_devices/native/oboe/src/common/Trace.h new file mode 100644 index 0000000000..c7965f956f --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/common/Trace.h @@ -0,0 +1,31 @@ +/* + * Copyright 2018 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_TRACE_H +#define OBOE_TRACE_H + +class Trace { + +public: + static void beginSection(const char *format, ...); + static void endSection(); + static void initialize(); + +private: + static bool mIsTracingSupported; +}; + +#endif //OBOE_TRACE_H \ No newline at end of file diff --git a/modules/juce_audio_devices/native/oboe/src/common/Utilities.cpp b/modules/juce_audio_devices/native/oboe/src/common/Utilities.cpp new file mode 100644 index 0000000000..f940d1bf68 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/common/Utilities.cpp @@ -0,0 +1,283 @@ +/* + * Copyright 2016 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + + +#include +#include +#include + +#ifdef __ANDROID__ +#include +#endif + +#include +#include "oboe/Definitions.h" +#include "oboe/Utilities.h" + +namespace oboe { + +constexpr float kScaleI16ToFloat = (1.0f / 32768.0f); + +void convertFloatToPcm16(const float *source, int16_t *destination, int32_t numSamples) { + for (int i = 0; i < numSamples; i++) { + float fval = source[i]; + fval += 1.0; // to avoid discontinuity at 0.0 caused by truncation + fval *= 32768.0f; + auto sample = static_cast(fval); + // clip to 16-bit range + if (sample < 0) sample = 0; + else if (sample > 0x0FFFF) sample = 0x0FFFF; + sample -= 32768; // center at zero + destination[i] = static_cast(sample); + } +} + +void convertPcm16ToFloat(const int16_t *source, float *destination, int32_t numSamples) { + for (int i = 0; i < numSamples; i++) { + destination[i] = source[i] * kScaleI16ToFloat; + } +} + +int32_t convertFormatToSizeInBytes(AudioFormat format) { + int32_t size = 0; + switch (format) { + case AudioFormat::I16: + size = sizeof(int16_t); + break; + case AudioFormat::Float: + size = sizeof(float); + break; + default: + break; + } + return size; +} + +template<> +const char *convertToText(Result returnCode) { + switch (returnCode) { + case Result::OK: return "OK"; + case Result::ErrorDisconnected: return "ErrorDisconnected"; + case Result::ErrorIllegalArgument: return "ErrorIllegalArgument"; + case Result::ErrorInternal: return "ErrorInternal"; + case Result::ErrorInvalidState: return "ErrorInvalidState"; + case Result::ErrorInvalidHandle: return "ErrorInvalidHandle"; + case Result::ErrorUnimplemented: return "ErrorUnimplemented"; + case Result::ErrorUnavailable: return "ErrorUnavailable"; + case Result::ErrorNoFreeHandles: return "ErrorNoFreeHandles"; + case Result::ErrorNoMemory: return "ErrorNoMemory"; + case Result::ErrorNull: return "ErrorNull"; + case Result::ErrorTimeout: return "ErrorTimeout"; + case Result::ErrorWouldBlock: return "ErrorWouldBlock"; + case Result::ErrorInvalidFormat: return "ErrorInvalidFormat"; + case Result::ErrorOutOfRange: return "ErrorOutOfRange"; + case Result::ErrorNoService: return "ErrorNoService"; + case Result::ErrorInvalidRate: return "ErrorInvalidRate"; + case Result::ErrorClosed: return "ErrorClosed"; + default: return "Unrecognized result"; + } +} + +template<> +const char *convertToText(AudioFormat format) { + switch (format) { + case AudioFormat::Invalid: return "Invalid"; + case AudioFormat::Unspecified: return "Unspecified"; + case AudioFormat::I16: return "I16"; + case AudioFormat::Float: return "Float"; + default: return "Unrecognized format"; + } +} + +template<> +const char *convertToText(PerformanceMode mode) { + switch (mode) { + case PerformanceMode::LowLatency: return "LowLatency"; + case PerformanceMode::None: return "None"; + case PerformanceMode::PowerSaving: return "PowerSaving"; + default: return "Unrecognized performance mode"; + } +} + +template<> +const char *convertToText(SharingMode mode) { + switch (mode) { + case SharingMode::Exclusive: return "Exclusive"; + case SharingMode::Shared: return "Shared"; + default: return "Unrecognized sharing mode"; + } +} + +template<> +const char *convertToText(DataCallbackResult result) { + switch (result) { + case DataCallbackResult::Continue: return "Continue"; + case DataCallbackResult::Stop: return "Stop"; + default: return "Unrecognized data callback result"; + } +} + +template<> +const char *convertToText(Direction direction) { + switch (direction) { + case Direction::Input: return "Input"; + case Direction::Output: return "Output"; + default: return "Unrecognized direction"; + } +} + +template<> +const char *convertToText(StreamState state) { + switch (state) { + case StreamState::Closed: return "Closed"; + case StreamState::Closing: return "Closing"; + case StreamState::Disconnected: return "Disconnected"; + case StreamState::Flushed: return "Flushed"; + case StreamState::Flushing: return "Flushing"; + case StreamState::Open: return "Open"; + case StreamState::Paused: return "Paused"; + case StreamState::Pausing: return "Pausing"; + case StreamState::Started: return "Started"; + case StreamState::Starting: return "Starting"; + case StreamState::Stopped: return "Stopped"; + case StreamState::Stopping: return "Stopping"; + case StreamState::Uninitialized: return "Uninitialized"; + case StreamState::Unknown: return "Unknown"; + default: return "Unrecognized stream state"; + } +} + +template<> +const char *convertToText(AudioApi audioApi) { + + switch (audioApi) { + case AudioApi::Unspecified: return "Unspecified"; + case AudioApi::OpenSLES: return "OpenSLES"; + case AudioApi::AAudio: return "AAudio"; + default: return "Unrecognized audio API"; + } +} + +template<> +const char *convertToText(AudioStream* stream) { + static std::string streamText; + std::stringstream s; + + s<<"StreamID: "<< static_cast(stream)<getDeviceId()<getDirection())<getAudioApi())<getBufferCapacityInFrames()<getBufferSizeInFrames()<getFramesPerBurst()<getFramesPerCallback()<getSampleRate()<getChannelCount()<getFormat())<getSharingMode())<getPerformanceMode()) + <getState())<getXRunCount()<getFramesRead()<getFramesWritten()< +const char *convertToText(Usage usage) { + + switch (usage) { + case Usage::Media: return "Media"; + case Usage::VoiceCommunication: return "VoiceCommunication"; + case Usage::VoiceCommunicationSignalling: return "VoiceCommunicationSignalling"; + case Usage::Alarm: return "Alarm"; + case Usage::Notification: return "Notification"; + case Usage::NotificationRingtone: return "NotificationRingtone"; + case Usage::NotificationEvent: return "NotificationEvent"; + case Usage::AssistanceAccessibility: return "AssistanceAccessibility"; + case Usage::AssistanceNavigationGuidance: return "AssistanceNavigationGuidance"; + case Usage::AssistanceSonification: return "AssistanceSonification"; + case Usage::Game: return "Game"; + case Usage::Assistant: return "Assistant"; + default: return "Unrecognized usage"; + } +} + +template<> +const char *convertToText(ContentType contentType) { + + switch (contentType) { + case ContentType::Speech: return "Speech"; + case ContentType::Music: return "Music"; + case ContentType::Movie: return "Movie"; + case ContentType::Sonification: return "Sonification"; + default: return "Unrecognized content type"; + } +} + +template<> +const char *convertToText(InputPreset inputPreset) { + + switch (inputPreset) { + case InputPreset::Generic: return "Generic"; + case InputPreset::Camcorder: return "Camcorder"; + case InputPreset::VoiceRecognition: return "VoiceRecognition"; + case InputPreset::VoiceCommunication: return "VoiceCommunication"; + case InputPreset::Unprocessed: return "Unprocessed"; + case InputPreset::VoicePerformance: return "VoicePerformance"; + default: return "Unrecognized input preset"; + } +} + +template<> +const char *convertToText(SessionId sessionId) { + + switch (sessionId) { + case SessionId::None: return "None"; + case SessionId::Allocate: return "Allocate"; + default: return "Unrecognized session id"; + } +} + +template<> +const char *convertToText(ChannelCount channelCount) { + + switch (channelCount) { + case ChannelCount::Unspecified: return "Unspecified"; + case ChannelCount::Mono: return "Mono"; + case ChannelCount::Stereo: return "Stereo"; + default: return "Unrecognized channel count"; + } +} + +int getSdkVersion() { +#ifdef __ANDROID__ + static int sCachedSdkVersion = -1; + if (sCachedSdkVersion == -1) { + char sdk[PROP_VALUE_MAX] = {0}; + if (__system_property_get("ro.build.version.sdk", sdk) != 0) { + sCachedSdkVersion = atoi(sdk); + } + } + return sCachedSdkVersion; +#endif + return -1; +} + +}// namespace oboe diff --git a/modules/juce_audio_devices/native/oboe/src/common/Version.cpp b/modules/juce_audio_devices/native/oboe/src/common/Version.cpp new file mode 100644 index 0000000000..481bee76f2 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/common/Version.cpp @@ -0,0 +1,28 @@ +/* + * Copyright 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ +#include "oboe/Version.h" + +namespace oboe { + + // This variable enables the version information to be read from the resulting binary e.g. + // by running `objdump -s --section=.data ` + // Please do not optimize or change in any way. + char kVersionText[] = "OboeVersion" OBOE_VERSION_TEXT; + + const char * getVersionText(){ + return kVersionText; + } +} // namespace oboe diff --git a/modules/juce_audio_devices/native/oboe/src/fifo/FifoBuffer.cpp b/modules/juce_audio_devices/native/oboe/src/fifo/FifoBuffer.cpp new file mode 100644 index 0000000000..f89dca6f11 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/fifo/FifoBuffer.cpp @@ -0,0 +1,186 @@ +/* + * Copyright 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include +#include +#include +#include +#include + +#include "common/OboeDebug.h" +#include "fifo/FifoControllerBase.h" +#include "fifo/FifoController.h" +#include "fifo/FifoControllerIndirect.h" +#include "fifo/FifoBuffer.h" +#include "common/AudioClock.h" + +namespace oboe { + +FifoBuffer::FifoBuffer(uint32_t bytesPerFrame, uint32_t capacityInFrames) + : mBytesPerFrame(bytesPerFrame) + , mStorage(nullptr) + , mFramesReadCount(0) + , mFramesUnderrunCount(0) +{ + mFifo = std::make_unique(capacityInFrames); + // allocate buffer + int32_t bytesPerBuffer = bytesPerFrame * capacityInFrames; + mStorage = new uint8_t[bytesPerBuffer]; + mStorageOwned = true; + LOGD("%s() capacityInFrames = %d, bytesPerFrame = %d", + __func__, capacityInFrames, bytesPerFrame); +} + +FifoBuffer::FifoBuffer( uint32_t bytesPerFrame, + uint32_t capacityInFrames, + std::atomic *readCounterAddress, + std::atomic *writeCounterAddress, + uint8_t *dataStorageAddress + ) + : mBytesPerFrame(bytesPerFrame) + , mStorage(dataStorageAddress) + , mFramesReadCount(0) + , mFramesUnderrunCount(0) +{ + mFifo = std::make_unique(capacityInFrames, + readCounterAddress, + writeCounterAddress); + mStorage = dataStorageAddress; + mStorageOwned = false; + LOGD("%s(*) capacityInFrames = %d, bytesPerFrame = %d", + __func__, capacityInFrames, bytesPerFrame); +} + +FifoBuffer::~FifoBuffer() { + if (mStorageOwned) { + delete[] mStorage; + } +} + +int32_t FifoBuffer::convertFramesToBytes(int32_t frames) { + return frames * mBytesPerFrame; +} + +int32_t FifoBuffer::read(void *buffer, int32_t numFrames) { + if (numFrames <= 0) { + return 0; + } + // safe because numFrames is guaranteed positive + uint32_t framesToRead = static_cast(numFrames); + uint32_t framesAvailable = mFifo->getFullFramesAvailable(); + framesToRead = std::min(framesToRead, framesAvailable); + + uint32_t readIndex = mFifo->getReadIndex(); // ranges 0 to capacity + uint8_t *destination = reinterpret_cast(buffer); + uint8_t *source = &mStorage[convertFramesToBytes(readIndex)]; + if ((readIndex + framesToRead) > mFifo->getFrameCapacity()) { + // read in two parts, first part here is at the end of the mStorage buffer + int32_t frames1 = static_cast(mFifo->getFrameCapacity() - readIndex); + int32_t numBytes = convertFramesToBytes(frames1); + if (numBytes < 0) { + return static_cast(Result::ErrorOutOfRange); + } + memcpy(destination, source, static_cast(numBytes)); + destination += numBytes; + // read second part, which is at the beginning of mStorage + source = &mStorage[0]; + int32_t frames2 = static_cast(framesToRead - frames1); + numBytes = convertFramesToBytes(frames2); + if (numBytes < 0) { + return static_cast(Result::ErrorOutOfRange); + } + memcpy(destination, source, static_cast(numBytes)); + } else { + // just read in one shot + int32_t numBytes = convertFramesToBytes(framesToRead); + if (numBytes < 0) { + return static_cast(Result::ErrorOutOfRange); + } + memcpy(destination, source, static_cast(numBytes)); + } + mFifo->advanceReadIndex(framesToRead); + + return framesToRead; +} + +int32_t FifoBuffer::write(const void *buffer, int32_t numFrames) { + if (numFrames <= 0) { + return 0; + } + // Guaranteed positive. + uint32_t framesToWrite = static_cast(numFrames); + uint32_t framesAvailable = mFifo->getEmptyFramesAvailable(); + framesToWrite = std::min(framesToWrite, framesAvailable); + + uint32_t writeIndex = mFifo->getWriteIndex(); + int byteIndex = convertFramesToBytes(writeIndex); + const uint8_t *source = reinterpret_cast(buffer); + uint8_t *destination = &mStorage[byteIndex]; + if ((writeIndex + framesToWrite) > mFifo->getFrameCapacity()) { + // write in two parts, first part here + int32_t frames1 = static_cast(mFifo->getFrameCapacity() - writeIndex); + int32_t numBytes = convertFramesToBytes(frames1); + if (numBytes < 0) { + return static_cast(Result::ErrorOutOfRange); + } + memcpy(destination, source, static_cast(numBytes)); + // read second part + source += convertFramesToBytes(frames1); + destination = &mStorage[0]; + int frames2 = static_cast(framesToWrite - frames1); + numBytes = convertFramesToBytes(frames2); + if (numBytes < 0) { + return static_cast(Result::ErrorOutOfRange); + } + memcpy(destination, source, static_cast(numBytes)); + } else { + // just write in one shot + int32_t numBytes = convertFramesToBytes(framesToWrite); + if (numBytes < 0) { + return static_cast(Result::ErrorOutOfRange); + } + memcpy(destination, source, static_cast(numBytes)); + } + mFifo->advanceWriteIndex(framesToWrite); + + return framesToWrite; +} + +int32_t FifoBuffer::readNow(void *buffer, int32_t numFrames) { + int32_t framesRead = read(buffer, numFrames); + if (framesRead < 0) { + return framesRead; + } + int32_t framesLeft = numFrames - framesRead; + mFramesReadCount += framesRead; + mFramesUnderrunCount += framesLeft; + // Zero out any samples we could not set. + if (framesLeft > 0) { + uint8_t *destination = reinterpret_cast(buffer); + destination += convertFramesToBytes(framesRead); // point to first byte not set + int32_t bytesToZero = convertFramesToBytes(framesLeft); + memset(destination, 0, static_cast(bytesToZero)); + } + + return framesRead; +} + + +uint32_t FifoBuffer::getBufferCapacityInFrames() const { + return mFifo->getFrameCapacity(); +} + +} // namespace oboe diff --git a/modules/juce_audio_devices/native/oboe/src/fifo/FifoBuffer.h b/modules/juce_audio_devices/native/oboe/src/fifo/FifoBuffer.h new file mode 100644 index 0000000000..80186208e6 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/fifo/FifoBuffer.h @@ -0,0 +1,99 @@ +/* + * Copyright 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_FIFOPROCESSOR_H +#define OBOE_FIFOPROCESSOR_H + +#include +#include + +#include "common/OboeDebug.h" +#include "FifoControllerBase.h" +#include "oboe/Definitions.h" + +namespace oboe { + +class FifoBuffer { +public: + FifoBuffer(uint32_t bytesPerFrame, uint32_t capacityInFrames); + + FifoBuffer(uint32_t bytesPerFrame, + uint32_t capacityInFrames, + std::atomic *readCounterAddress, + std::atomic *writeCounterAddress, + uint8_t *dataStorageAddress); + + ~FifoBuffer(); + + int32_t convertFramesToBytes(int32_t frames); + + /** + * Read framesToRead or, if not enough, then read as many as are available. + * @param destination + * @param framesToRead number of frames requested + * @return number of frames actually read + */ + int32_t read(void *destination, int32_t framesToRead); + + int32_t write(const void *source, int32_t framesToWrite); + + uint32_t getBufferCapacityInFrames() const; + + /** + * Calls read(). If all of the frames cannot be read then the remainder of the buffer + * is set to zero. + * + * @param destination + * @param framesToRead number of frames requested + * @return number of frames actually read + */ + int32_t readNow(void *destination, int32_t numFrames); + + uint32_t getFullFramesAvailable() { + return mFifo->getFullFramesAvailable(); + } + + uint32_t getBytesPerFrame() const { + return mBytesPerFrame; + } + + uint64_t getReadCounter() const { + return mFifo->getReadCounter(); + } + + void setReadCounter(uint64_t n) { + mFifo->setReadCounter(n); + } + + uint64_t getWriteCounter() { + return mFifo->getWriteCounter(); + } + void setWriteCounter(uint64_t n) { + mFifo->setWriteCounter(n); + } + +private: + uint32_t mBytesPerFrame; + uint8_t* mStorage; + bool mStorageOwned; // did this object allocate the storage? + std::unique_ptr mFifo; + uint64_t mFramesReadCount; + uint64_t mFramesUnderrunCount; +}; + +} // namespace oboe + +#endif //OBOE_FIFOPROCESSOR_H diff --git a/modules/juce_audio_devices/native/oboe/src/fifo/FifoController.cpp b/modules/juce_audio_devices/native/oboe/src/fifo/FifoController.cpp new file mode 100644 index 0000000000..5590683ba4 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/fifo/FifoController.cpp @@ -0,0 +1,31 @@ +/* + * Copyright 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include +#include +#include "FifoControllerBase.h" +#include "FifoController.h" + +namespace oboe { + +FifoController::FifoController(uint32_t numFrames) + : FifoControllerBase(numFrames) +{ + setReadCounter(0); + setWriteCounter(0); +} + +} // namespace oboe diff --git a/modules/juce_audio_devices/native/oboe/src/fifo/FifoController.h b/modules/juce_audio_devices/native/oboe/src/fifo/FifoController.h new file mode 100644 index 0000000000..6562e6dec2 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/fifo/FifoController.h @@ -0,0 +1,61 @@ +/* + * Copyright 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef NATIVEOBOE_FIFOCONTROLLER_H +#define NATIVEOBOE_FIFOCONTROLLER_H + +#include +#include "FifoControllerBase.h" +#include + +namespace oboe { + +/** + * A FifoControllerBase with counters contained in the class. + */ +class FifoController : public FifoControllerBase +{ +public: + FifoController(uint32_t bufferSize); + virtual ~FifoController() = default; + + virtual uint64_t getReadCounter() const override { + return mReadCounter.load(std::memory_order_acquire); + } + virtual void setReadCounter(uint64_t n) override { + mReadCounter.store(n, std::memory_order_release); + } + virtual void incrementReadCounter(uint64_t n) override { + mReadCounter.fetch_add(n, std::memory_order_acq_rel); + } + virtual uint64_t getWriteCounter() const override { + return mWriteCounter.load(std::memory_order_acquire); + } + virtual void setWriteCounter(uint64_t n) override { + mWriteCounter.store(n, std::memory_order_release); + } + virtual void incrementWriteCounter(uint64_t n) override { + mWriteCounter.fetch_add(n, std::memory_order_acq_rel); + } + +private: + std::atomic mReadCounter{}; + std::atomic mWriteCounter{}; +}; + +} // namespace oboe + +#endif //NATIVEOBOE_FIFOCONTROLLER_H diff --git a/modules/juce_audio_devices/native/oboe/src/fifo/FifoControllerBase.cpp b/modules/juce_audio_devices/native/oboe/src/fifo/FifoControllerBase.cpp new file mode 100644 index 0000000000..6aceea0e81 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/fifo/FifoControllerBase.cpp @@ -0,0 +1,71 @@ +/* + * Copyright 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include "FifoControllerBase.h" + +#include +#include +#include +#include "FifoControllerBase.h" + +#include "common/OboeDebug.h" + +namespace oboe { + +FifoControllerBase::FifoControllerBase(uint32_t capacityInFrames) + : mTotalFrames(capacityInFrames) +{ + // Avoid ridiculously large buffers and the arithmetic wraparound issues that can follow. + assert(capacityInFrames <= (UINT32_MAX / 4)); +} + +uint32_t FifoControllerBase::getFullFramesAvailable() const { + uint64_t writeCounter = getWriteCounter(); + uint64_t readCounter = getReadCounter(); + if (readCounter > writeCounter) { + return 0; + } + uint64_t delta = writeCounter - readCounter; + if (delta >= mTotalFrames) { + return mTotalFrames; + } + // delta is now guaranteed to fit within the range of a uint32_t + return static_cast(delta); +} + +uint32_t FifoControllerBase::getReadIndex() const { + // % works with non-power of two sizes + return static_cast(getReadCounter() % mTotalFrames); +} + +void FifoControllerBase::advanceReadIndex(uint32_t numFrames) { + incrementReadCounter(numFrames); +} + +uint32_t FifoControllerBase::getEmptyFramesAvailable() const { + return static_cast(mTotalFrames - getFullFramesAvailable()); +} + +uint32_t FifoControllerBase::getWriteIndex() const { + // % works with non-power of two sizes + return static_cast(getWriteCounter() % mTotalFrames); +} + +void FifoControllerBase::advanceWriteIndex(uint32_t numFrames) { + incrementWriteCounter(numFrames); +} + +} // namespace oboe diff --git a/modules/juce_audio_devices/native/oboe/src/fifo/FifoControllerBase.h b/modules/juce_audio_devices/native/oboe/src/fifo/FifoControllerBase.h new file mode 100644 index 0000000000..c8041e041b --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/fifo/FifoControllerBase.h @@ -0,0 +1,93 @@ +/* + * Copyright 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef NATIVEOBOE_FIFOCONTROLLERBASE_H +#define NATIVEOBOE_FIFOCONTROLLERBASE_H + +#include +#include + +namespace oboe { + +/** + * Manage the read/write indices of a circular buffer. + * + * The caller is responsible for reading and writing the actual data. + * Note that the span of available frames may not be contiguous. They + * may wrap around from the end to the beginning of the buffer. In that + * case the data must be read or written in at least two blocks of frames. + * + */ + +class FifoControllerBase { + +public: + /** + * @param totalFrames capacity of the circular buffer in frames. + */ + FifoControllerBase(uint32_t totalFrames); + + virtual ~FifoControllerBase() = default; + + /** + * The frames available to read will be calculated from the read and write counters. + * The result will be clipped to the capacity of the buffer. + * If the buffer has underflowed then this will return zero. + * @return number of valid frames available to read. + */ + uint32_t getFullFramesAvailable() const; + + /** + * The index in a circular buffer of the next frame to read. + */ + uint32_t getReadIndex() const; + + /** + * @param numFrames number of frames to advance the read index + */ + void advanceReadIndex(uint32_t numFrames); + + /** + * @return maximum number of frames that can be written without exceeding the threshold. + */ + uint32_t getEmptyFramesAvailable() const; + + /** + * The index in a circular buffer of the next frame to write. + */ + uint32_t getWriteIndex() const; + + /** + * @param numFrames number of frames to advance the write index + */ + void advanceWriteIndex(uint32_t numFrames); + + uint32_t getFrameCapacity() const { return mTotalFrames; } + + virtual uint64_t getReadCounter() const = 0; + virtual void setReadCounter(uint64_t n) = 0; + virtual void incrementReadCounter(uint64_t n) = 0; + virtual uint64_t getWriteCounter() const = 0; + virtual void setWriteCounter(uint64_t n) = 0; + virtual void incrementWriteCounter(uint64_t n) = 0; + +private: + uint32_t mTotalFrames; +}; + +} // namespace oboe + +#endif //NATIVEOBOE_FIFOCONTROLLERBASE_H diff --git a/modules/juce_audio_devices/native/oboe/src/fifo/FifoControllerIndirect.cpp b/modules/juce_audio_devices/native/oboe/src/fifo/FifoControllerIndirect.cpp new file mode 100644 index 0000000000..f596461fbd --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/fifo/FifoControllerIndirect.cpp @@ -0,0 +1,31 @@ +/* + * Copyright 2016 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + + +#include "FifoControllerIndirect.h" + +namespace oboe { + +FifoControllerIndirect::FifoControllerIndirect(uint32_t numFrames, + std::atomic *readCounterAddress, + std::atomic *writeCounterAddress) + : FifoControllerBase(numFrames) + , mReadCounterAddress(readCounterAddress) + , mWriteCounterAddress(writeCounterAddress) +{ +} + +} diff --git a/modules/juce_audio_devices/native/oboe/src/fifo/FifoControllerIndirect.h b/modules/juce_audio_devices/native/oboe/src/fifo/FifoControllerIndirect.h new file mode 100644 index 0000000000..216a28b536 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/fifo/FifoControllerIndirect.h @@ -0,0 +1,64 @@ +/* + * Copyright 2016 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef NATIVEOBOE_FIFOCONTROLLERINDIRECT_H +#define NATIVEOBOE_FIFOCONTROLLERINDIRECT_H + +#include "FifoControllerBase.h" +#include + +namespace oboe { + +/** + * A FifoControllerBase with counters external to the class. + */ +class FifoControllerIndirect : public FifoControllerBase { + +public: + FifoControllerIndirect(uint32_t bufferSize, + std::atomic *readCounterAddress, + std::atomic *writeCounterAddress); + virtual ~FifoControllerIndirect() = default; + + virtual uint64_t getReadCounter() const override { + return mReadCounterAddress->load(std::memory_order_acquire); + } + virtual void setReadCounter(uint64_t n) override { + mReadCounterAddress->store(n, std::memory_order_release); + } + virtual void incrementReadCounter(uint64_t n) override { + mReadCounterAddress->fetch_add(n, std::memory_order_acq_rel); + } + virtual uint64_t getWriteCounter() const override { + return mWriteCounterAddress->load(std::memory_order_acquire); + } + virtual void setWriteCounter(uint64_t n) override { + mWriteCounterAddress->store(n, std::memory_order_release); + } + virtual void incrementWriteCounter(uint64_t n) override { + mWriteCounterAddress->fetch_add(n, std::memory_order_acq_rel); + } + +private: + + std::atomic *mReadCounterAddress; + std::atomic *mWriteCounterAddress; + +}; + +} // namespace oboe + +#endif //NATIVEOBOE_FIFOCONTROLLERINDIRECT_H diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/ClipToRange.cpp b/modules/juce_audio_devices/native/oboe/src/flowgraph/ClipToRange.cpp new file mode 100644 index 0000000000..d2f8a02acd --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/ClipToRange.cpp @@ -0,0 +1,38 @@ +/* + * Copyright 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include +#include +#include "FlowGraphNode.h" +#include "ClipToRange.h" + +using namespace flowgraph; + +ClipToRange::ClipToRange(int32_t channelCount) + : FlowGraphFilter(channelCount) { +} + +int32_t ClipToRange::onProcess(int32_t numFrames) { + const float *inputBuffer = input.getBuffer(); + float *outputBuffer = output.getBuffer(); + + int32_t numSamples = numFrames * output.getSamplesPerFrame(); + for (int32_t i = 0; i < numSamples; i++) { + *outputBuffer++ = std::min(mMaximum, std::max(mMinimum, *inputBuffer++)); + } + + return numFrames; +} diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/ClipToRange.h b/modules/juce_audio_devices/native/oboe/src/flowgraph/ClipToRange.h new file mode 100644 index 0000000000..22b780496e --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/ClipToRange.h @@ -0,0 +1,68 @@ +/* + * Copyright 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef FLOWGRAPH_CLIP_TO_RANGE_H +#define FLOWGRAPH_CLIP_TO_RANGE_H + +#include +#include +#include + +#include "FlowGraphNode.h" + +namespace flowgraph { + +// This is 3 dB, (10^(3/20)), to match the maximum headroom in AudioTrack for float data. +// It is designed to allow occasional transient peaks. +constexpr float kDefaultMaxHeadroom = 1.41253754f; +constexpr float kDefaultMinHeadroom = -kDefaultMaxHeadroom; + +class ClipToRange : public FlowGraphFilter { +public: + explicit ClipToRange(int32_t channelCount); + + virtual ~ClipToRange() = default; + + int32_t onProcess(int32_t numFrames) override; + + void setMinimum(float min) { + mMinimum = min; + } + + float getMinimum() const { + return mMinimum; + } + + void setMaximum(float min) { + mMaximum = min; + } + + float getMaximum() const { + return mMaximum; + } + + const char *getName() override { + return "ClipToRange"; + } + +private: + float mMinimum = kDefaultMinHeadroom; + float mMaximum = kDefaultMaxHeadroom; +}; + +} /* namespace flowgraph */ + +#endif //FLOWGRAPH_CLIP_TO_RANGE_H diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/FlowGraphNode.cpp b/modules/juce_audio_devices/native/oboe/src/flowgraph/FlowGraphNode.cpp new file mode 100644 index 0000000000..bb6ecc9fa4 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/FlowGraphNode.cpp @@ -0,0 +1,111 @@ +/* + * Copyright 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include "stdio.h" +#include +#include +#include "FlowGraphNode.h" + +using namespace flowgraph; + +/***************************************************************************/ +int32_t FlowGraphNode::pullData(int64_t framePosition, int32_t numFrames) { + int32_t frameCount = numFrames; + // Prevent recursion and multiple execution of nodes. + if (framePosition <= mLastFramePosition && !mBlockRecursion) { + mBlockRecursion = true; // for cyclic graphs + if (mDataPulledAutomatically) { + // Pull from all the upstream nodes. + for (auto &port : mInputPorts) { + // TODO fix bug of leaving unused data in some ports if using multiple AudioSource + frameCount = port.get().pullData(framePosition, frameCount); + } + } + if (frameCount > 0) { + frameCount = onProcess(frameCount); + } + mLastFramePosition += frameCount; + mBlockRecursion = false; + mLastFrameCount = frameCount; + } else { + frameCount = mLastFrameCount; + } + return frameCount; +} + +void FlowGraphNode::pullReset() { + if (!mBlockRecursion) { + mBlockRecursion = true; // for cyclic graphs + // Pull reset from all the upstream nodes. + for (auto &port : mInputPorts) { + port.get().pullReset(); + } + mBlockRecursion = false; + reset(); + } +} + +void FlowGraphNode::reset() { + mLastFrameCount = 0; +} + +/***************************************************************************/ +FlowGraphPortFloat::FlowGraphPortFloat(FlowGraphNode &parent, + int32_t samplesPerFrame, + int32_t framesPerBuffer) + : FlowGraphPort(parent, samplesPerFrame) + , mFramesPerBuffer(framesPerBuffer) + , mBuffer(nullptr) { + size_t numFloats = static_cast(framesPerBuffer * getSamplesPerFrame()); + mBuffer = std::make_unique(numFloats); +} + +/***************************************************************************/ +int32_t FlowGraphPortFloatOutput::pullData(int64_t framePosition, int32_t numFrames) { + numFrames = std::min(getFramesPerBuffer(), numFrames); + return mContainingNode.pullData(framePosition, numFrames); +} + +void FlowGraphPortFloatOutput::pullReset() { + mContainingNode.pullReset(); +} + +// These need to be in the .cpp file because of forward cross references. +void FlowGraphPortFloatOutput::connect(FlowGraphPortFloatInput *port) { + port->connect(this); +} + +void FlowGraphPortFloatOutput::disconnect(FlowGraphPortFloatInput *port) { + port->disconnect(this); +} + +/***************************************************************************/ +int32_t FlowGraphPortFloatInput::pullData(int64_t framePosition, int32_t numFrames) { + return (mConnected == nullptr) + ? std::min(getFramesPerBuffer(), numFrames) + : mConnected->pullData(framePosition, numFrames); +} +void FlowGraphPortFloatInput::pullReset() { + if (mConnected != nullptr) mConnected->pullReset(); +} + +float *FlowGraphPortFloatInput::getBuffer() { + if (mConnected == nullptr) { + return FlowGraphPortFloat::getBuffer(); // loaded using setValue() + } else { + return mConnected->getBuffer(); + } +} diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/FlowGraphNode.h b/modules/juce_audio_devices/native/oboe/src/flowgraph/FlowGraphNode.h new file mode 100644 index 0000000000..007131e47c --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/FlowGraphNode.h @@ -0,0 +1,422 @@ +/* + * Copyright 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +/* + * FlowGraph.h + * + * Processing node and ports that can be used in a simple data flow graph. + * This was designed to work with audio but could be used for other + * types of data. + */ + +#ifndef FLOWGRAPH_FLOW_GRAPH_NODE_H +#define FLOWGRAPH_FLOW_GRAPH_NODE_H + +#include +#include +#include +#include +#include +#include +#include +#include + +// TODO Move these classes into separate files. +// TODO Review use of raw pointers for connect(). Maybe use smart pointers but need to avoid +// run-time deallocation in audio thread. + +// Set this to 1 if using it inside the Android framework. +// This code is kept here so that it can be moved easily between Oboe and AAudio. +#define FLOWGRAPH_ANDROID_INTERNAL 0 + +namespace flowgraph { + +// Default block size that can be overridden when the FlowGraphPortFloat is created. +// If it is too small then we will have too much overhead from switching between nodes. +// If it is too high then we will thrash the caches. +constexpr int kDefaultBufferSize = 8; // arbitrary + +class FlowGraphPort; +class FlowGraphPortFloatInput; + +/***************************************************************************/ +/** + * Base class for all nodes in the flowgraph. + */ +class FlowGraphNode { +public: + FlowGraphNode() {} + virtual ~FlowGraphNode() = default; + + /** + * Read from the input ports, + * generate multiple frames of data then write the results to the output ports. + * + * @param numFrames maximum number of frames requested for processing + * @return number of frames actually processed + */ + virtual int32_t onProcess(int32_t numFrames) = 0; + + /** + * If the framePosition is at or after the last frame position then call onProcess(). + * This prevents infinite recursion in case of cyclic graphs. + * It also prevents nodes upstream from a branch from being executed twice. + * + * @param framePosition + * @param numFrames + * @return number of frames valid + */ + int32_t pullData(int64_t framePosition, int32_t numFrames); + + /** + * Recursively reset all the nodes in the graph, starting from a Sink. + * + * This must not be called at the same time as pullData! + */ + void pullReset(); + + /** + * Reset framePosition counters. + */ + virtual void reset(); + + void addInputPort(FlowGraphPort &port) { + mInputPorts.push_back(port); + } + + bool isDataPulledAutomatically() const { + return mDataPulledAutomatically; + } + + /** + * Set true if you want the data pulled through the graph automatically. + * This is the default. + * + * Set false if you want to pull the data from the input ports in the onProcess() method. + * You might do this, for example, in a sample rate converting node. + * + * @param automatic + */ + void setDataPulledAutomatically(bool automatic) { + mDataPulledAutomatically = automatic; + } + + virtual const char *getName() { + return "FlowGraph"; + } + + int64_t getLastFramePosition() { + return mLastFramePosition; + } + +protected: + int64_t mLastFramePosition = 0; + + std::vector> mInputPorts; + +private: + bool mDataPulledAutomatically = true; + bool mBlockRecursion = false; + int32_t mLastFrameCount = 0; + +}; + +/***************************************************************************/ +/** + * This is a connector that allows data to flow between modules. + * + * The ports are the primary means of interacting with a module. + * So they are generally declared as public. + * + */ +class FlowGraphPort { +public: + FlowGraphPort(FlowGraphNode &parent, int32_t samplesPerFrame) + : mContainingNode(parent) + , mSamplesPerFrame(samplesPerFrame) { + } + + // Ports are often declared public. So let's make them non-copyable. + FlowGraphPort(const FlowGraphPort&) = delete; + FlowGraphPort& operator=(const FlowGraphPort&) = delete; + + int32_t getSamplesPerFrame() const { + return mSamplesPerFrame; + } + + virtual int32_t pullData(int64_t framePosition, int32_t numFrames) = 0; + + virtual void pullReset() {} + +protected: + FlowGraphNode &mContainingNode; + +private: + const int32_t mSamplesPerFrame = 1; +}; + +/***************************************************************************/ +/** + * This port contains a 32-bit float buffer that can contain several frames of data. + * Processing the data in a block improves performance. + * + * The size is framesPerBuffer * samplesPerFrame). + */ +class FlowGraphPortFloat : public FlowGraphPort { +public: + FlowGraphPortFloat(FlowGraphNode &parent, + int32_t samplesPerFrame, + int32_t framesPerBuffer = kDefaultBufferSize + ); + + virtual ~FlowGraphPortFloat() = default; + + int32_t getFramesPerBuffer() const { + return mFramesPerBuffer; + } + +protected: + + /** + * @return buffer internal to the port or from a connected port + */ + virtual float *getBuffer() { + return mBuffer.get(); + } + +private: + const int32_t mFramesPerBuffer = 1; + std::unique_ptr mBuffer; // allocated in constructor +}; + +/***************************************************************************/ +/** + * The results of a node's processing are stored in the buffers of the output ports. + */ +class FlowGraphPortFloatOutput : public FlowGraphPortFloat { +public: + FlowGraphPortFloatOutput(FlowGraphNode &parent, int32_t samplesPerFrame) + : FlowGraphPortFloat(parent, samplesPerFrame) { + } + + virtual ~FlowGraphPortFloatOutput() = default; + + using FlowGraphPortFloat::getBuffer; + + /** + * Connect to the input of another module. + * An input port can only have one connection. + * An output port can have multiple connections. + * If you connect a second output port to an input port + * then it overwrites the previous connection. + * + * This not thread safe. Do not modify the graph topology from another thread while running. + * Also do not delete a module while it is connected to another port if the graph is running. + */ + void connect(FlowGraphPortFloatInput *port); + + /** + * Disconnect from the input of another module. + * This not thread safe. + */ + void disconnect(FlowGraphPortFloatInput *port); + + /** + * Call the parent module's onProcess() method. + * That may pull data from its inputs and recursively + * process the entire graph. + * @return number of frames actually pulled + */ + int32_t pullData(int64_t framePosition, int32_t numFrames) override; + + + void pullReset() override; + +}; + +/***************************************************************************/ + +/** + * An input port for streaming audio data. + * You can set a value that will be used for processing. + * If you connect an output port to this port then its value will be used instead. + */ +class FlowGraphPortFloatInput : public FlowGraphPortFloat { +public: + FlowGraphPortFloatInput(FlowGraphNode &parent, int32_t samplesPerFrame) + : FlowGraphPortFloat(parent, samplesPerFrame) { + // Add to parent so it can pull data from each input. + parent.addInputPort(*this); + } + + virtual ~FlowGraphPortFloatInput() = default; + + /** + * If connected to an output port then this will return + * that output ports buffers. + * If not connected then it returns the input ports own buffer + * which can be loaded using setValue(). + */ + float *getBuffer() override; + + /** + * Write every value of the float buffer. + * This value will be ignored if an output port is connected + * to this port. + */ + void setValue(float value) { + int numFloats = kDefaultBufferSize * getSamplesPerFrame(); + float *buffer = getBuffer(); + for (int i = 0; i < numFloats; i++) { + *buffer++ = value; + } + } + + /** + * Connect to the output of another module. + * An input port can only have one connection. + * An output port can have multiple connections. + * This not thread safe. + */ + void connect(FlowGraphPortFloatOutput *port) { + assert(getSamplesPerFrame() == port->getSamplesPerFrame()); + mConnected = port; + } + + void disconnect(FlowGraphPortFloatOutput *port) { + assert(mConnected == port); + (void) port; + mConnected = nullptr; + } + + void disconnect() { + mConnected = nullptr; + } + + /** + * Pull data from any output port that is connected. + */ + int32_t pullData(int64_t framePosition, int32_t numFrames) override; + + void pullReset() override; + +private: + FlowGraphPortFloatOutput *mConnected = nullptr; +}; + +/***************************************************************************/ + +/** + * Base class for an edge node in a graph that has no upstream nodes. + * It outputs data but does not consume data. + * By default, it will read its data from an external buffer. + */ +class FlowGraphSource : public FlowGraphNode { +public: + explicit FlowGraphSource(int32_t channelCount) + : output(*this, channelCount) { + } + + virtual ~FlowGraphSource() = default; + + FlowGraphPortFloatOutput output; +}; + +/***************************************************************************/ + +/** + * Base class for an edge node in a graph that has no upstream nodes. + * It outputs data but does not consume data. + * By default, it will read its data from an external buffer. + */ +class FlowGraphSourceBuffered : public FlowGraphSource { +public: + explicit FlowGraphSourceBuffered(int32_t channelCount) + : FlowGraphSource(channelCount) {} + + virtual ~FlowGraphSourceBuffered() = default; + + /** + * Specify buffer that the node will read from. + * + * @param data TODO Consider using std::shared_ptr. + * @param numFrames + */ + void setData(const void *data, int32_t numFrames) { + mData = data; + mSizeInFrames = numFrames; + mFrameIndex = 0; + } + +protected: + const void *mData = nullptr; + int32_t mSizeInFrames = 0; // number of frames in mData + int32_t mFrameIndex = 0; // index of next frame to be processed +}; + +/***************************************************************************/ +/** + * Base class for an edge node in a graph that has no downstream nodes. + * It consumes data but does not output data. + * This graph will be executed when data is read() from this node + * by pulling data from upstream nodes. + */ +class FlowGraphSink : public FlowGraphNode { +public: + explicit FlowGraphSink(int32_t channelCount) + : input(*this, channelCount) { + } + + virtual ~FlowGraphSink() = default; + + FlowGraphPortFloatInput input; + + /** + * Dummy processor. The work happens in the read() method. + * + * @param numFrames + * @return number of frames actually processed + */ + int32_t onProcess(int32_t numFrames) override { + return numFrames; + } + + virtual int32_t read(int64_t framePosition, void *data, int32_t numFrames) = 0; + +}; + +/***************************************************************************/ +/** + * Base class for a node that has an input and an output with the same number of channels. + * This may include traditional filters, eg. FIR, but also include + * any processing node that converts input to output. + */ +class FlowGraphFilter : public FlowGraphNode { +public: + explicit FlowGraphFilter(int32_t channelCount) + : input(*this, channelCount) + , output(*this, channelCount) { + } + + virtual ~FlowGraphFilter() = default; + + FlowGraphPortFloatInput input; + FlowGraphPortFloatOutput output; +}; + +} /* namespace flowgraph */ + +#endif /* FLOWGRAPH_FLOW_GRAPH_NODE_H */ diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/ManyToMultiConverter.cpp b/modules/juce_audio_devices/native/oboe/src/flowgraph/ManyToMultiConverter.cpp new file mode 100644 index 0000000000..879685e87c --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/ManyToMultiConverter.cpp @@ -0,0 +1,47 @@ +/* + * Copyright 2018 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include + +#include "ManyToMultiConverter.h" + +using namespace flowgraph; + +ManyToMultiConverter::ManyToMultiConverter(int32_t channelCount) + : inputs(channelCount) + , output(*this, channelCount) { + for (int i = 0; i < channelCount; i++) { + inputs[i] = std::make_unique(*this, 1); + } +} + +int32_t ManyToMultiConverter::onProcess(int32_t numFrames) { + int32_t channelCount = output.getSamplesPerFrame(); + + for (int ch = 0; ch < channelCount; ch++) { + const float *inputBuffer = inputs[ch]->getBuffer(); + float *outputBuffer = output.getBuffer() + ch; + + for (int i = 0; i < numFrames; i++) { + // read one, write into the proper interleaved output channel + float sample = *inputBuffer++; + *outputBuffer = sample; + outputBuffer += channelCount; // advance to next multichannel frame + } + } + return numFrames; +} + diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/ManyToMultiConverter.h b/modules/juce_audio_devices/native/oboe/src/flowgraph/ManyToMultiConverter.h new file mode 100644 index 0000000000..eca4a8ed71 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/ManyToMultiConverter.h @@ -0,0 +1,49 @@ +/* + * Copyright 2018 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef FLOWGRAPH_MANY_TO_MULTI_CONVERTER_H +#define FLOWGRAPH_MANY_TO_MULTI_CONVERTER_H + +#include +#include +#include + +#include "FlowGraphNode.h" + +/** + * Combine multiple mono inputs into one interleaved multi-channel output. + */ +class ManyToMultiConverter : public flowgraph::FlowGraphNode { +public: + explicit ManyToMultiConverter(int32_t channelCount); + + virtual ~ManyToMultiConverter() = default; + + int32_t onProcess(int numFrames) override; + + void setEnabled(bool enabled) {} + + std::vector> inputs; + flowgraph::FlowGraphPortFloatOutput output; + + const char *getName() override { + return "ManyToMultiConverter"; + } + +private: +}; + +#endif //FLOWGRAPH_MANY_TO_MULTI_CONVERTER_H diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/MonoToMultiConverter.cpp b/modules/juce_audio_devices/native/oboe/src/flowgraph/MonoToMultiConverter.cpp new file mode 100644 index 0000000000..11cea78883 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/MonoToMultiConverter.cpp @@ -0,0 +1,43 @@ +/* + * Copyright 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include +#include "FlowGraphNode.h" +#include "MonoToMultiConverter.h" + +using namespace flowgraph; + +MonoToMultiConverter::MonoToMultiConverter(int32_t channelCount) + : input(*this, 1) + , output(*this, channelCount) { +} + +MonoToMultiConverter::~MonoToMultiConverter() { } + +int32_t MonoToMultiConverter::onProcess(int32_t numFrames) { + const float *inputBuffer = input.getBuffer(); + float *outputBuffer = output.getBuffer(); + int32_t channelCount = output.getSamplesPerFrame(); + for (int i = 0; i < numFrames; i++) { + // read one, write many + float sample = *inputBuffer++; + for (int channel = 0; channel < channelCount; channel++) { + *outputBuffer++ = sample; + } + } + return numFrames; +} + diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/MonoToMultiConverter.h b/modules/juce_audio_devices/native/oboe/src/flowgraph/MonoToMultiConverter.h new file mode 100644 index 0000000000..376f7a37fa --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/MonoToMultiConverter.h @@ -0,0 +1,49 @@ +/* + * Copyright 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef FLOWGRAPH_MONO_TO_MULTI_CONVERTER_H +#define FLOWGRAPH_MONO_TO_MULTI_CONVERTER_H + +#include +#include + +#include "FlowGraphNode.h" + +namespace flowgraph { + +/** + * Convert a monophonic stream to a multi-channel stream + * with the same signal on each channel. + */ +class MonoToMultiConverter : public FlowGraphNode { +public: + explicit MonoToMultiConverter(int32_t channelCount); + + virtual ~MonoToMultiConverter(); + + int32_t onProcess(int32_t numFrames) override; + + const char *getName() override { + return "MonoToMultiConverter"; + } + + FlowGraphPortFloatInput input; + FlowGraphPortFloatOutput output; +}; + +} /* namespace flowgraph */ + +#endif //FLOWGRAPH_MONO_TO_MULTI_CONVERTER_H diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/RampLinear.cpp b/modules/juce_audio_devices/native/oboe/src/flowgraph/RampLinear.cpp new file mode 100644 index 0000000000..afef0181cb --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/RampLinear.cpp @@ -0,0 +1,77 @@ +/* + * Copyright 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include +#include +#include "FlowGraphNode.h" +#include "RampLinear.h" + +using namespace flowgraph; + +RampLinear::RampLinear(int32_t channelCount) + : FlowGraphFilter(channelCount) { + mTarget.store(1.0f); +} + +void RampLinear::setLengthInFrames(int32_t frames) { + mLengthInFrames = frames; +} + +void RampLinear::setTarget(float target) { + mTarget.store(target); +} + +float RampLinear::interpolateCurrent() { + return mLevelTo - (mRemaining * mScaler); +} + +int32_t RampLinear::onProcess(int32_t numFrames) { + const float *inputBuffer = input.getBuffer(); + float *outputBuffer = output.getBuffer(); + int32_t channelCount = output.getSamplesPerFrame(); + + float target = getTarget(); + if (target != mLevelTo) { + // Start new ramp. Continue from previous level. + mLevelFrom = interpolateCurrent(); + mLevelTo = target; + mRemaining = mLengthInFrames; + mScaler = (mLevelTo - mLevelFrom) / mLengthInFrames; // for interpolation + } + + int32_t framesLeft = numFrames; + + if (mRemaining > 0) { // Ramping? This doesn't happen very often. + int32_t framesToRamp = std::min(framesLeft, mRemaining); + framesLeft -= framesToRamp; + while (framesToRamp > 0) { + float currentLevel = interpolateCurrent(); + for (int ch = 0; ch < channelCount; ch++) { + *outputBuffer++ = *inputBuffer++ * currentLevel; + } + mRemaining--; + framesToRamp--; + } + } + + // Process any frames after the ramp. + int32_t samplesLeft = framesLeft * channelCount; + for (int i = 0; i < samplesLeft; i++) { + *outputBuffer++ = *inputBuffer++ * mLevelTo; + } + + return numFrames; +} diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/RampLinear.h b/modules/juce_audio_devices/native/oboe/src/flowgraph/RampLinear.h new file mode 100644 index 0000000000..f285704c73 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/RampLinear.h @@ -0,0 +1,96 @@ +/* + * Copyright 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef FLOWGRAPH_RAMP_LINEAR_H +#define FLOWGRAPH_RAMP_LINEAR_H + +#include +#include +#include + +#include "FlowGraphNode.h" + +namespace flowgraph { + +/** + * When the target is modified then the output will ramp smoothly + * between the original and the new target value. + * This can be used to smooth out control values and reduce pops. + * + * The target may be updated while a ramp is in progress, which will trigger + * a new ramp from the current value. + */ +class RampLinear : public FlowGraphFilter { +public: + explicit RampLinear(int32_t channelCount); + + virtual ~RampLinear() = default; + + int32_t onProcess(int32_t numFrames) override; + + /** + * This is used for the next ramp. + * Calling this does not affect a ramp that is in progress. + */ + void setLengthInFrames(int32_t frames); + + int32_t getLengthInFrames() const { + return mLengthInFrames; + } + + /** + * This may be safely called by another thread. + * @param target + */ + void setTarget(float target); + + float getTarget() const { + return mTarget.load(); + } + + /** + * Force the nextSegment to start from this level. + * + * WARNING: this can cause a discontinuity if called while the ramp is being used. + * Only call this when setting the initial ramp. + * + * @param level + */ + void forceCurrent(float level) { + mLevelFrom = level; + mLevelTo = level; + } + + const char *getName() override { + return "RampLinear"; + } + +private: + + float interpolateCurrent(); + + std::atomic mTarget; + + int32_t mLengthInFrames = 48000.0f / 100.0f ; // 10 msec at 48000 Hz; + int32_t mRemaining = 0; + float mScaler = 0.0f; + float mLevelFrom = 0.0f; + float mLevelTo = 0.0f; +}; + +} /* namespace flowgraph */ + +#endif //FLOWGRAPH_RAMP_LINEAR_H diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/SampleRateConverter.cpp b/modules/juce_audio_devices/native/oboe/src/flowgraph/SampleRateConverter.cpp new file mode 100644 index 0000000000..708c684fc3 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/SampleRateConverter.cpp @@ -0,0 +1,64 @@ +/* + * Copyright 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include "SampleRateConverter.h" + +using namespace flowgraph; +using namespace resampler; + +SampleRateConverter::SampleRateConverter(int32_t channelCount, MultiChannelResampler &resampler) + : FlowGraphFilter(channelCount) + , mResampler(resampler) { + setDataPulledAutomatically(false); +} + +// Return true if there is a sample available. +bool SampleRateConverter::isInputAvailable() { + if (mInputCursor >= mNumValidInputFrames) { + mNumValidInputFrames = input.pullData(mInputFramePosition, input.getFramesPerBuffer()); + mInputFramePosition += mNumValidInputFrames; + mInputCursor = 0; + } + return (mInputCursor < mNumValidInputFrames); +} + +const float *SampleRateConverter::getNextInputFrame() { + const float *inputBuffer = input.getBuffer(); + return &inputBuffer[mInputCursor++ * input.getSamplesPerFrame()]; +} + +int32_t SampleRateConverter::onProcess(int32_t numFrames) { + float *outputBuffer = output.getBuffer(); + int32_t channelCount = output.getSamplesPerFrame(); + int framesLeft = numFrames; + while (framesLeft > 0) { + // Gather input samples as needed. + if(mResampler.isWriteNeeded()) { + if (isInputAvailable()) { + const float *frame = getNextInputFrame(); + mResampler.writeNextFrame(frame); + } else { + break; + } + } else { + // Output frame is interpolated from input samples. + mResampler.readNextFrame(outputBuffer); + outputBuffer += channelCount; + framesLeft--; + } + } + return numFrames - framesLeft; +} diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/SampleRateConverter.h b/modules/juce_audio_devices/native/oboe/src/flowgraph/SampleRateConverter.h new file mode 100644 index 0000000000..d940b22b98 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/SampleRateConverter.h @@ -0,0 +1,56 @@ +/* + * Copyright 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_SAMPLE_RATE_CONVERTER_H +#define OBOE_SAMPLE_RATE_CONVERTER_H + +#include +#include + +#include "FlowGraphNode.h" +#include "resampler/MultiChannelResampler.h" + +namespace flowgraph { + +class SampleRateConverter : public FlowGraphFilter { +public: + explicit SampleRateConverter(int32_t channelCount, resampler::MultiChannelResampler &mResampler); + + virtual ~SampleRateConverter() = default; + + int32_t onProcess(int32_t numFrames) override; + + const char *getName() override { + return "SampleRateConverter"; + } + +private: + + // Return true if there is a sample available. + bool isInputAvailable(); + + // This assumes data is available. Only call after calling isInputAvailable(). + const float *getNextInputFrame(); + + resampler::MultiChannelResampler &mResampler; + + int32_t mInputCursor = 0; + int32_t mNumValidInputFrames = 0; + int64_t mInputFramePosition = 0; // monotonic counter of input frames used for pullData + +}; +} /* namespace flowgraph */ +#endif //OBOE_SAMPLE_RATE_CONVERTER_H diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/SinkFloat.cpp b/modules/juce_audio_devices/native/oboe/src/flowgraph/SinkFloat.cpp new file mode 100644 index 0000000000..f830dafd2b --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/SinkFloat.cpp @@ -0,0 +1,50 @@ +/* + * Copyright 2018 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include +#include +#include "FlowGraphNode.h" +#include "SinkFloat.h" + +using namespace flowgraph; + +SinkFloat::SinkFloat(int32_t channelCount) + : FlowGraphSink(channelCount) { +} + +int32_t SinkFloat::read(int64_t framePosition, void *data, int32_t numFrames) { + // printf("SinkFloat::read(,,%d)\n", numFrames); + float *floatData = (float *) data; + int32_t channelCount = input.getSamplesPerFrame(); + + int32_t framesLeft = numFrames; + while (framesLeft > 0) { + // Run the graph and pull data through the input port. + int32_t framesPulled = pullData(framePosition, framesLeft); + // printf("SinkFloat::read: framesLeft = %d, framesPulled = %d\n", framesLeft, framesPulled); + if (framesPulled <= 0) { + break; + } + const float *signal = input.getBuffer(); + int32_t numSamples = framesPulled * channelCount; + memcpy(floatData, signal, numSamples * sizeof(float)); + floatData += numSamples; + framesLeft -= framesPulled; + framePosition += framesPulled; + } + // printf("SinkFloat returning %d\n", numFrames - framesLeft); + return numFrames - framesLeft; +} diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/SinkFloat.h b/modules/juce_audio_devices/native/oboe/src/flowgraph/SinkFloat.h new file mode 100644 index 0000000000..b6474d1802 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/SinkFloat.h @@ -0,0 +1,44 @@ +/* + * Copyright 2018 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + + +#ifndef FLOWGRAPH_SINK_FLOAT_H +#define FLOWGRAPH_SINK_FLOAT_H + +#include +#include + +#include "FlowGraphNode.h" + +namespace flowgraph { + +/** + * AudioSink that lets you read data as 32-bit floats. + */ +class SinkFloat : public FlowGraphSink { +public: + explicit SinkFloat(int32_t channelCount); + + int32_t read(int64_t framePosition, void *data, int32_t numFrames) override; + + const char *getName() override { + return "SinkFloat"; + } +}; + +} /* namespace flowgraph */ + +#endif //FLOWGRAPH_SINK_FLOAT_H diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/SinkI16.cpp b/modules/juce_audio_devices/native/oboe/src/flowgraph/SinkI16.cpp new file mode 100644 index 0000000000..a5904e82fa --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/SinkI16.cpp @@ -0,0 +1,58 @@ +/* + * Copyright 2018 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include +#include + +#include "SinkI16.h" + +#if FLOWGRAPH_ANDROID_INTERNAL +#include +#endif + +using namespace flowgraph; + +SinkI16::SinkI16(int32_t channelCount) + : FlowGraphSink(channelCount) {} + +int32_t SinkI16::read(int64_t framePosition, void *data, int32_t numFrames) { + int16_t *shortData = (int16_t *) data; + const int32_t channelCount = input.getSamplesPerFrame(); + + int32_t framesLeft = numFrames; + while (framesLeft > 0) { + // Run the graph and pull data through the input port. + int32_t framesRead = pullData(framePosition, framesLeft); + if (framesRead <= 0) { + break; + } + const float *signal = input.getBuffer(); + int32_t numSamples = framesRead * channelCount; +#if FLOWGRAPH_ANDROID_INTERNAL + memcpy_to_i16_from_float(shortData, signal, numSamples); + shortData += numSamples; + signal += numSamples; +#else + for (int i = 0; i < numSamples; i++) { + int32_t n = (int32_t) (*signal++ * 32768.0f); + *shortData++ = std::min(INT16_MAX, std::max(INT16_MIN, n)); // clip + } +#endif + framesLeft -= framesRead; + framePosition += framesRead; + } + return numFrames - framesLeft; +} diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/SinkI16.h b/modules/juce_audio_devices/native/oboe/src/flowgraph/SinkI16.h new file mode 100644 index 0000000000..2cfdfb0cb6 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/SinkI16.h @@ -0,0 +1,43 @@ +/* + * Copyright 2018 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef FLOWGRAPH_SINK_I16_H +#define FLOWGRAPH_SINK_I16_H + +#include +#include + +#include "FlowGraphNode.h" + +namespace flowgraph { + +/** + * AudioSink that lets you read data as 16-bit signed integers. + */ +class SinkI16 : public FlowGraphSink { +public: + explicit SinkI16(int32_t channelCount); + + int32_t read(int64_t framePosition, void *data, int32_t numFrames) override; + + const char *getName() override { + return "SinkI16"; + } +}; + +} /* namespace flowgraph */ + +#endif //FLOWGRAPH_SINK_I16_H diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/SinkI24.cpp b/modules/juce_audio_devices/native/oboe/src/flowgraph/SinkI24.cpp new file mode 100644 index 0000000000..b944b770b5 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/SinkI24.cpp @@ -0,0 +1,67 @@ +/* + * Copyright 2018 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include +#include + + +#include "FlowGraphNode.h" +#include "SinkI24.h" + +#if FLOWGRAPH_ANDROID_INTERNAL +#include +#endif + +using namespace flowgraph; + +SinkI24::SinkI24(int32_t channelCount) + : FlowGraphSink(channelCount) {} + +int32_t SinkI24::read(int64_t framePosition, void *data, int32_t numFrames) { + uint8_t *byteData = (uint8_t *) data; + const int32_t channelCount = input.getSamplesPerFrame(); + + int32_t framesLeft = numFrames; + while (framesLeft > 0) { + // Run the graph and pull data through the input port. + int32_t framesRead = pullData(framePosition, framesLeft); + if (framesRead <= 0) { + break; + } + const float *floatData = input.getBuffer(); + int32_t numSamples = framesRead * channelCount; +#if FLOWGRAPH_ANDROID_INTERNAL + memcpy_to_p24_from_float(byteData, floatData, numSamples); + static const int kBytesPerI24Packed = 3; + byteData += numSamples * kBytesPerI24Packed; + floatData += numSamples; +#else + const int32_t kI24PackedMax = 0x007FFFFF; + const int32_t kI24PackedMin = 0xFF800000; + for (int i = 0; i < numSamples; i++) { + int32_t n = (int32_t) (*floatData++ * 0x00800000); + n = std::min(kI24PackedMax, std::max(kI24PackedMin, n)); // clip + // Write as a packed 24-bit integer in Little Endian format. + *byteData++ = (uint8_t) n; + *byteData++ = (uint8_t) (n >> 8); + *byteData++ = (uint8_t) (n >> 16); + } +#endif + framesLeft -= framesRead; + framePosition += framesRead; + } + return numFrames - framesLeft; +} diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/SinkI24.h b/modules/juce_audio_devices/native/oboe/src/flowgraph/SinkI24.h new file mode 100644 index 0000000000..7477c8d29a --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/SinkI24.h @@ -0,0 +1,44 @@ +/* + * Copyright 2018 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef FLOWGRAPH_SINK_I24_H +#define FLOWGRAPH_SINK_I24_H + +#include +#include + +#include "FlowGraphNode.h" + +namespace flowgraph { + +/** + * AudioSink that lets you read data as packed 24-bit signed integers. + * The sample size is 3 bytes. + */ +class SinkI24 : public FlowGraphSink { +public: + explicit SinkI24(int32_t channelCount); + + int32_t read(int64_t framePosition, void *data, int32_t numFrames) override; + + const char *getName() override { + return "SinkI24"; + } +}; + +} /* namespace flowgraph */ + +#endif //FLOWGRAPH_SINK_I24_H diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/SourceFloat.cpp b/modules/juce_audio_devices/native/oboe/src/flowgraph/SourceFloat.cpp new file mode 100644 index 0000000000..f574d84203 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/SourceFloat.cpp @@ -0,0 +1,43 @@ +/* + * Copyright 2018 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include "common/OboeDebug.h" +#include +#include +#include "FlowGraphNode.h" +#include "SourceFloat.h" + +using namespace flowgraph; + +SourceFloat::SourceFloat(int32_t channelCount) + : FlowGraphSourceBuffered(channelCount) { +} + +int32_t SourceFloat::onProcess(int32_t numFrames) { + float *outputBuffer = output.getBuffer(); + int32_t channelCount = output.getSamplesPerFrame(); + + int32_t framesLeft = mSizeInFrames - mFrameIndex; + int32_t framesToProcess = std::min(numFrames, framesLeft); + int32_t numSamples = framesToProcess * channelCount; + + const float *floatBase = (float *) mData; + const float *floatData = &floatBase[mFrameIndex * channelCount]; + memcpy(outputBuffer, floatData, numSamples * sizeof(float)); + mFrameIndex += framesToProcess; + return framesToProcess; +} + diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/SourceFloat.h b/modules/juce_audio_devices/native/oboe/src/flowgraph/SourceFloat.h new file mode 100644 index 0000000000..4de1b41c35 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/SourceFloat.h @@ -0,0 +1,43 @@ +/* + * Copyright 2018 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef FLOWGRAPH_SOURCE_FLOAT_H +#define FLOWGRAPH_SOURCE_FLOAT_H + +#include +#include + +#include "FlowGraphNode.h" + +namespace flowgraph { + +/** + * AudioSource that reads a block of pre-defined float data. + */ +class SourceFloat : public FlowGraphSourceBuffered { +public: + explicit SourceFloat(int32_t channelCount); + + int32_t onProcess(int32_t numFrames) override; + + const char *getName() override { + return "SourceFloat"; + } +}; + +} /* namespace flowgraph */ + +#endif //FLOWGRAPH_SOURCE_FLOAT_H diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/SourceI16.cpp b/modules/juce_audio_devices/native/oboe/src/flowgraph/SourceI16.cpp new file mode 100644 index 0000000000..881302390b --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/SourceI16.cpp @@ -0,0 +1,54 @@ +/* + * Copyright 2018 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include +#include + +#include "FlowGraphNode.h" +#include "SourceI16.h" + +#if FLOWGRAPH_ANDROID_INTERNAL +#include +#endif + +using namespace flowgraph; + +SourceI16::SourceI16(int32_t channelCount) + : FlowGraphSourceBuffered(channelCount) { +} + +int32_t SourceI16::onProcess(int32_t numFrames) { + float *floatData = output.getBuffer(); + int32_t channelCount = output.getSamplesPerFrame(); + + int32_t framesLeft = mSizeInFrames - mFrameIndex; + int32_t framesToProcess = std::min(numFrames, framesLeft); + int32_t numSamples = framesToProcess * channelCount; + + const int16_t *shortBase = static_cast(mData); + const int16_t *shortData = &shortBase[mFrameIndex * channelCount]; + +#if FLOWGRAPH_ANDROID_INTERNAL + memcpy_to_float_from_i16(floatData, shortData, numSamples); +#else + for (int i = 0; i < numSamples; i++) { + *floatData++ = *shortData++ * (1.0f / 32768); + } +#endif + + mFrameIndex += framesToProcess; + return framesToProcess; +} \ No newline at end of file diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/SourceI16.h b/modules/juce_audio_devices/native/oboe/src/flowgraph/SourceI16.h new file mode 100644 index 0000000000..fe440b2006 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/SourceI16.h @@ -0,0 +1,42 @@ +/* + * Copyright 2018 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef FLOWGRAPH_SOURCE_I16_H +#define FLOWGRAPH_SOURCE_I16_H + +#include +#include + +#include "FlowGraphNode.h" + +namespace flowgraph { +/** + * AudioSource that reads a block of pre-defined 16-bit integer data. + */ +class SourceI16 : public FlowGraphSourceBuffered { +public: + explicit SourceI16(int32_t channelCount); + + int32_t onProcess(int32_t numFrames) override; + + const char *getName() override { + return "SourceI16"; + } +}; + +} /* namespace flowgraph */ + +#endif //FLOWGRAPH_SOURCE_I16_H diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/SourceI24.cpp b/modules/juce_audio_devices/native/oboe/src/flowgraph/SourceI24.cpp new file mode 100644 index 0000000000..1975878799 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/SourceI24.cpp @@ -0,0 +1,65 @@ +/* + * Copyright 2018 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include +#include + +#if FLOWGRAPH_ANDROID_INTERNAL +#include +#endif + +#include "FlowGraphNode.h" +#include "SourceI24.h" + +using namespace flowgraph; + +constexpr int kBytesPerI24Packed = 3; + +SourceI24::SourceI24(int32_t channelCount) + : FlowGraphSourceBuffered(channelCount) { +} + +int32_t SourceI24::onProcess(int32_t numFrames) { + float *floatData = output.getBuffer(); + int32_t channelCount = output.getSamplesPerFrame(); + + int32_t framesLeft = mSizeInFrames - mFrameIndex; + int32_t framesToProcess = std::min(numFrames, framesLeft); + int32_t numSamples = framesToProcess * channelCount; + + const uint8_t *byteBase = (uint8_t *) mData; + const uint8_t *byteData = &byteBase[mFrameIndex * channelCount * kBytesPerI24Packed]; + +#if FLOWGRAPH_ANDROID_INTERNAL + memcpy_to_float_from_p24(floatData, byteData, numSamples); +#else + static const float scale = 1. / (float)(1UL << 31); + for (int i = 0; i < numSamples; i++) { + // Assemble the data assuming Little Endian format. + int32_t pad = byteData[2]; + pad <<= 8; + pad |= byteData[1]; + pad <<= 8; + pad |= byteData[0]; + pad <<= 8; // Shift to 32 bit data so the sign is correct. + byteData += kBytesPerI24Packed; + *floatData++ = pad * scale; // scale to range -1.0 to 1.0 + } +#endif + + mFrameIndex += framesToProcess; + return framesToProcess; +} \ No newline at end of file diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/SourceI24.h b/modules/juce_audio_devices/native/oboe/src/flowgraph/SourceI24.h new file mode 100644 index 0000000000..377953432f --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/SourceI24.h @@ -0,0 +1,43 @@ +/* + * Copyright 2018 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef FLOWGRAPH_SOURCE_I24_H +#define FLOWGRAPH_SOURCE_I24_H + +#include +#include + +#include "FlowGraphNode.h" + +namespace flowgraph { + +/** + * AudioSource that reads a block of pre-defined 24-bit packed integer data. + */ +class SourceI24 : public FlowGraphSourceBuffered { +public: + explicit SourceI24(int32_t channelCount); + + int32_t onProcess(int32_t numFrames) override; + + const char *getName() override { + return "SourceI24"; + } +}; + +} /* namespace flowgraph */ + +#endif //FLOWGRAPH_SOURCE_I24_H diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/HyperbolicCosineWindow.h b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/HyperbolicCosineWindow.h new file mode 100644 index 0000000000..f6479ae327 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/HyperbolicCosineWindow.h @@ -0,0 +1,68 @@ +/* + * Copyright 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef RESAMPLER_HYPERBOLIC_COSINE_WINDOW_H +#define RESAMPLER_HYPERBOLIC_COSINE_WINDOW_H + +#include + +namespace resampler { + +/** + * Calculate a HyperbolicCosineWindow window centered at 0. + * This can be used in place of a Kaiser window. + * + * The code is based on an anonymous contribution by "a concerned citizen": + * https://dsp.stackexchange.com/questions/37714/kaiser-window-approximation + */ +class HyperbolicCosineWindow { +public: + HyperbolicCosineWindow() { + setStopBandAttenuation(60); + } + + /** + * @param attenuation typical values range from 30 to 90 dB + * @return beta + */ + double setStopBandAttenuation(double attenuation) { + double alpha = ((-325.1e-6 * attenuation + 0.1677) * attenuation) - 3.149; + setAlpha(alpha); + return alpha; + } + + void setAlpha(double alpha) { + mAlpha = alpha; + mInverseCoshAlpha = 1.0 / cosh(alpha); + } + + /** + * @param x ranges from -1.0 to +1.0 + */ + double operator()(double x) { + double x2 = x * x; + if (x2 >= 1.0) return 0.0; + double w = mAlpha * sqrt(1.0 - x2); + return cosh(w) * mInverseCoshAlpha; + } + +private: + double mAlpha = 0.0; + double mInverseCoshAlpha = 1.0; +}; + +} // namespace resampler +#endif //RESAMPLER_HYPERBOLIC_COSINE_WINDOW_H diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/IntegerRatio.cpp b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/IntegerRatio.cpp new file mode 100644 index 0000000000..d2f51801dc --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/IntegerRatio.cpp @@ -0,0 +1,52 @@ +/* + * Copyright 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include + +#include "IntegerRatio.h" + +using namespace resampler; + +// Enough primes to cover the common sample rates. +const std::vector IntegerRatio::kPrimes{ + 2, 3, 5, 7, 11, 13, 17, 19, 23, 29, 31, 37, 41, + 43, 47, 53, 59, 61, 67, 71, 73, 79, 83, 89, 97, + 101, 103, 107, 109, 113, 127, 131, 137, 139, 149, + 151, 157, 163, 167, 173, 179, 181, 191, 193, 197, 199}; + +void IntegerRatio::reduce() { + for (int prime : kPrimes) { + if (mNumerator < prime || mDenominator < prime) { + break; + } + + // Find biggest prime factor for numerator. + while (true) { + int top = mNumerator / prime; + int bottom = mDenominator / prime; + if ((top >= 1) + && (bottom >= 1) + && (top * prime == mNumerator) // divided evenly? + && (bottom * prime == mDenominator)) { + mNumerator = top; + mDenominator = bottom; + } else { + break; + } + } + + } +} diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/IntegerRatio.h b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/IntegerRatio.h new file mode 100644 index 0000000000..d0b3a48d9a --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/IntegerRatio.h @@ -0,0 +1,54 @@ +/* + * Copyright 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_INTEGER_RATIO_H +#define OBOE_INTEGER_RATIO_H + +#include +#include + +namespace resampler { + +/** + * Represent the ratio of two integers. + */ +class IntegerRatio { +public: + IntegerRatio(int32_t numerator, int32_t denominator) + : mNumerator(numerator), mDenominator(denominator) {} + + /** + * Reduce by removing common prime factors. + */ + void reduce(); + + int32_t getNumerator() { + return mNumerator; + } + + int32_t getDenominator() { + return mDenominator; + } + +private: + int32_t mNumerator; + int32_t mDenominator; + static const std::vector kPrimes; +}; + +} + +#endif //OBOE_INTEGER_RATIO_H diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/KaiserWindow.h b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/KaiserWindow.h new file mode 100644 index 0000000000..73dbc41785 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/KaiserWindow.h @@ -0,0 +1,87 @@ +/* + * Copyright 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef RESAMPLER_KAISER_WINDOW_H +#define RESAMPLER_KAISER_WINDOW_H + +#include + +namespace resampler { + +/** + * Calculate a Kaiser window centered at 0. + */ +class KaiserWindow { +public: + KaiserWindow() { + setStopBandAttenuation(60); + } + + /** + * @param attenuation typical values range from 30 to 90 dB + * @return beta + */ + double setStopBandAttenuation(double attenuation) { + double beta = 0.0; + if (attenuation > 50) { + beta = 0.1102 * (attenuation - 8.7); + } else if (attenuation >= 21) { + double a21 = attenuation - 21; + beta = 0.5842 * pow(a21, 0.4) + (0.07886 * a21); + } + setBeta(beta); + return beta; + } + + void setBeta(double beta) { + mBeta = beta; + mInverseBesselBeta = 1.0 / bessel(beta); + } + + /** + * @param x ranges from -1.0 to +1.0 + */ + double operator()(double x) { + double x2 = x * x; + if (x2 >= 1.0) return 0.0; + double w = mBeta * sqrt(1.0 - x2); + return bessel(w) * mInverseBesselBeta; + } + + // Approximation of a + // modified zero order Bessel function of the first kind. + // Based on a discussion at: + // https://dsp.stackexchange.com/questions/37714/kaiser-window-approximation + static double bessel(double x) { + double y = cosh(0.970941817426052 * x); + y += cosh(0.8854560256532099 * x); + y += cosh(0.7485107481711011 * x); + y += cosh(0.5680647467311558 * x); + y += cosh(0.3546048870425356 * x); + y += cosh(0.120536680255323 * x); + y *= 2; + y += cosh(x); + y /= 13; + return y; + } + +private: + double mBeta = 0.0; + double mInverseBesselBeta = 1.0; +}; + +} // namespace resampler +#endif //RESAMPLER_KAISER_WINDOW_H diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/LinearResampler.cpp b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/LinearResampler.cpp new file mode 100644 index 0000000000..a7748c10fd --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/LinearResampler.cpp @@ -0,0 +1,42 @@ +/* + * Copyright 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include "LinearResampler.h" + +using namespace resampler; + +LinearResampler::LinearResampler(const MultiChannelResampler::Builder &builder) + : MultiChannelResampler(builder) { + mPreviousFrame = std::make_unique(getChannelCount()); + mCurrentFrame = std::make_unique(getChannelCount()); +} + +void LinearResampler::writeFrame(const float *frame) { + memcpy(mPreviousFrame.get(), mCurrentFrame.get(), sizeof(float) * getChannelCount()); + memcpy(mCurrentFrame.get(), frame, sizeof(float) * getChannelCount()); +} + +void LinearResampler::readFrame(float *frame) { + float *previous = mPreviousFrame.get(); + float *current = mCurrentFrame.get(); + float phase = (float) getIntegerPhase() / mDenominator; + // iterate across samples in the frame + for (int channel = 0; channel < getChannelCount(); channel++) { + float f0 = *previous++; + float f1 = *current++; + *frame++ = f0 + (phase * (f1 - f0)); + } +} diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/LinearResampler.h b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/LinearResampler.h new file mode 100644 index 0000000000..5dcc88130d --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/LinearResampler.h @@ -0,0 +1,44 @@ +/* + * Copyright 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_LINEAR_RESAMPLER_H +#define OBOE_LINEAR_RESAMPLER_H + +#include +#include +#include +#include "MultiChannelResampler.h" + +namespace resampler { + +/** + * Simple resampler that uses bi-linear interpolation. + */ +class LinearResampler : public MultiChannelResampler { +public: + LinearResampler(const MultiChannelResampler::Builder &builder); + + void writeFrame(const float *frame) override; + + void readFrame(float *frame) override; + +private: + std::unique_ptr mPreviousFrame; + std::unique_ptr mCurrentFrame; +}; + +} // namespace resampler +#endif //OBOE_LINEAR_RESAMPLER_H diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/MultiChannelResampler.cpp b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/MultiChannelResampler.cpp new file mode 100644 index 0000000000..d63052077e --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/MultiChannelResampler.cpp @@ -0,0 +1,171 @@ +/* + * Copyright 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include + +#include "IntegerRatio.h" +#include "LinearResampler.h" +#include "MultiChannelResampler.h" +#include "PolyphaseResampler.h" +#include "PolyphaseResamplerMono.h" +#include "PolyphaseResamplerStereo.h" +#include "SincResampler.h" +#include "SincResamplerStereo.h" + +using namespace resampler; + +MultiChannelResampler::MultiChannelResampler(const MultiChannelResampler::Builder &builder) + : mNumTaps(builder.getNumTaps()) + , mX(builder.getChannelCount() * builder.getNumTaps() * 2) + , mSingleFrame(builder.getChannelCount()) + , mChannelCount(builder.getChannelCount()) + { + // Reduce sample rates to the smallest ratio. + // For example 44100/48000 would become 147/160. + IntegerRatio ratio(builder.getInputRate(), builder.getOutputRate()); + ratio.reduce(); + mNumerator = ratio.getNumerator(); + mDenominator = ratio.getDenominator(); + mIntegerPhase = mDenominator; +} + +// static factory method +MultiChannelResampler *MultiChannelResampler::make(int32_t channelCount, + int32_t inputRate, + int32_t outputRate, + Quality quality) { + Builder builder; + builder.setInputRate(inputRate); + builder.setOutputRate(outputRate); + builder.setChannelCount(channelCount); + + switch (quality) { + case Quality::Fastest: + builder.setNumTaps(2); + break; + case Quality::Low: + builder.setNumTaps(4); + break; + case Quality::Medium: + default: + builder.setNumTaps(8); + break; + case Quality::High: + builder.setNumTaps(16); + break; + case Quality::Best: + builder.setNumTaps(32); + break; + } + + // Set the cutoff frequency so that we do not get aliasing when down-sampling. + if (inputRate > outputRate) { + builder.setNormalizedCutoff(kDefaultNormalizedCutoff); + } + return builder.build(); +} + +MultiChannelResampler *MultiChannelResampler::Builder::build() { + if (getNumTaps() == 2) { + // Note that this does not do low pass filteringh. + return new LinearResampler(*this); + } + IntegerRatio ratio(getInputRate(), getOutputRate()); + ratio.reduce(); + bool usePolyphase = (getNumTaps() * ratio.getDenominator()) <= kMaxCoefficients; + if (usePolyphase) { + if (getChannelCount() == 1) { + return new PolyphaseResamplerMono(*this); + } else if (getChannelCount() == 2) { + return new PolyphaseResamplerStereo(*this); + } else { + return new PolyphaseResampler(*this); + } + } else { + // Use less optimized resampler that uses a float phaseIncrement. + // TODO mono resampler + if (getChannelCount() == 2) { + return new SincResamplerStereo(*this); + } else { + return new SincResampler(*this); + } + } +} + +void MultiChannelResampler::writeFrame(const float *frame) { + // Move cursor before write so that cursor points to last written frame in read. + if (--mCursor < 0) { + mCursor = getNumTaps() - 1; + } + float *dest = &mX[mCursor * getChannelCount()]; + int offset = getNumTaps() * getChannelCount(); + for (int channel = 0; channel < getChannelCount(); channel++) { + // Write twice so we avoid having to wrap when reading. + dest[channel] = dest[channel + offset] = frame[channel]; + } +} + +float MultiChannelResampler::sinc(float radians) { + if (abs(radians) < 1.0e-9) return 1.0f; // avoid divide by zero + return sinf(radians) / radians; // Sinc function +} + +// Generate coefficients in the order they will be used by readFrame(). +// This is more complicated but readFrame() is called repeatedly and should be optimized. +void MultiChannelResampler::generateCoefficients(int32_t inputRate, + int32_t outputRate, + int32_t numRows, + double phaseIncrement, + float normalizedCutoff) { + mCoefficients.resize(getNumTaps() * numRows); + int coefficientIndex = 0; + double phase = 0.0; // ranges from 0.0 to 1.0, fraction between samples + // Stretch the sinc function for low pass filtering. + const float cutoffScaler = normalizedCutoff * + ((outputRate < inputRate) + ? ((float)outputRate / inputRate) + : ((float)inputRate / outputRate)); + const int numTapsHalf = getNumTaps() / 2; // numTaps must be even. + const float numTapsHalfInverse = 1.0f / numTapsHalf; + for (int i = 0; i < numRows; i++) { + float tapPhase = phase - numTapsHalf; + float gain = 0.0; // sum of raw coefficients + int gainCursor = coefficientIndex; + for (int tap = 0; tap < getNumTaps(); tap++) { + float radians = tapPhase * M_PI; + +#if MCR_USE_KAISER + float window = mKaiserWindow(tapPhase * numTapsHalfInverse); +#else + float window = mCoshWindow(tapPhase * numTapsHalfInverse); +#endif + float coefficient = sinc(radians * cutoffScaler) * window; + mCoefficients.at(coefficientIndex++) = coefficient; + gain += coefficient; + tapPhase += 1.0; + } + phase += phaseIncrement; + while (phase >= 1.0) { + phase -= 1.0; + } + + // Correct for gain variations. + float gainCorrection = 1.0 / gain; // normalize the gain + for (int tap = 0; tap < getNumTaps(); tap++) { + mCoefficients.at(gainCursor + tap) *= gainCorrection; + } + } +} diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/MultiChannelResampler.h b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/MultiChannelResampler.h new file mode 100644 index 0000000000..8b23d810d4 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/MultiChannelResampler.h @@ -0,0 +1,271 @@ +/* + * Copyright 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_MULTICHANNEL_RESAMPLER_H +#define OBOE_MULTICHANNEL_RESAMPLER_H + +#include +#include +#include +#include + +#ifndef MCR_USE_KAISER +// It appears from the spectrogram that the HyperbolicCosine window leads to fewer artifacts. +// And it is faster to calculate. +#define MCR_USE_KAISER 0 +#endif + +#if MCR_USE_KAISER +#include "KaiserWindow.h" +#else +#include "HyperbolicCosineWindow.h" +#endif + +namespace resampler { + +class MultiChannelResampler { + +public: + + enum class Quality : int32_t { + Fastest, + Low, + Medium, + High, + Best, + }; + + class Builder { + public: + /** + * Construct an optimal resampler based on the specified parameters. + * @return address of a resampler + */ + MultiChannelResampler *build(); + + /** + * The number of taps in the resampling filter. + * More taps gives better quality but uses more CPU time. + * This typically ranges from 4 to 64. Default is 16. + * + * For polyphase filters, numTaps must be a multiple of four for loop unrolling. + * @param numTaps number of taps for the filter + * @return address of this builder for chaining calls + */ + Builder *setNumTaps(int32_t numTaps) { + mNumTaps = numTaps; + return this; + } + + /** + * Use 1 for mono, 2 for stereo, etc. Default is 1. + * + * @param channelCount number of channels + * @return address of this builder for chaining calls + */ + Builder *setChannelCount(int32_t channelCount) { + mChannelCount = channelCount; + return this; + } + + /** + * Default is 48000. + * + * @param inputRate sample rate of the input stream + * @return address of this builder for chaining calls + */ + Builder *setInputRate(int32_t inputRate) { + mInputRate = inputRate; + return this; + } + + /** + * Default is 48000. + * + * @param outputRate sample rate of the output stream + * @return address of this builder for chaining calls + */ + Builder *setOutputRate(int32_t outputRate) { + mOutputRate = outputRate; + return this; + } + + /** + * Set cutoff frequency relative to the Nyquist rate of the output sample rate. + * Set to 1.0 to match the Nyquist frequency. + * Set lower to reduce aliasing. + * Default is 0.70. + * + * @param normalizedCutoff anti-aliasing filter cutoff + * @return address of this builder for chaining calls + */ + Builder *setNormalizedCutoff(float normalizedCutoff) { + mNormalizedCutoff = normalizedCutoff; + return this; + } + + int32_t getNumTaps() const { + return mNumTaps; + } + + int32_t getChannelCount() const { + return mChannelCount; + } + + int32_t getInputRate() const { + return mInputRate; + } + + int32_t getOutputRate() const { + return mOutputRate; + } + + float getNormalizedCutoff() const { + return mNormalizedCutoff; + } + + protected: + int32_t mChannelCount = 1; + int32_t mNumTaps = 16; + int32_t mInputRate = 48000; + int32_t mOutputRate = 48000; + float mNormalizedCutoff = kDefaultNormalizedCutoff; + }; + + virtual ~MultiChannelResampler() = default; + + /** + * Factory method for making a resampler that is optimal for the given inputs. + * + * @param channelCount number of channels, 2 for stereo + * @param inputRate sample rate of the input stream + * @param outputRate sample rate of the output stream + * @param quality higher quality sounds better but uses more CPU + * @return an optimal resampler + */ + static MultiChannelResampler *make(int32_t channelCount, + int32_t inputRate, + int32_t outputRate, + Quality quality); + + bool isWriteNeeded() const { + return mIntegerPhase >= mDenominator; + } + + /** + * Write a frame containing N samples. + * + * @param frame pointer to the first sample in a frame + */ + void writeNextFrame(const float *frame) { + writeFrame(frame); + advanceWrite(); + } + + /** + * Read a frame containing N samples. + * + * @param frame pointer to the first sample in a frame + */ + void readNextFrame(float *frame) { + readFrame(frame); + advanceRead(); + } + + int getNumTaps() const { + return mNumTaps; + } + + int getChannelCount() const { + return mChannelCount; + } + + static float hammingWindow(float radians, float spread); + + static float sinc(float radians); + +protected: + + explicit MultiChannelResampler(const MultiChannelResampler::Builder &builder); + + /** + * Write a frame containing N samples. + * Call advanceWrite() after calling this. + * @param frame pointer to the first sample in a frame + */ + virtual void writeFrame(const float *frame); + + /** + * Read a frame containing N samples using interpolation. + * Call advanceRead() after calling this. + * @param frame pointer to the first sample in a frame + */ + virtual void readFrame(float *frame) = 0; + + void advanceWrite() { + mIntegerPhase -= mDenominator; + } + + void advanceRead() { + mIntegerPhase += mNumerator; + } + + /** + * Generate the filter coefficients in optimal order. + * @param inputRate sample rate of the input stream + * @param outputRate sample rate of the output stream + * @param numRows number of rows in the array that contain a set of tap coefficients + * @param phaseIncrement how much to increment the phase between rows + * @param normalizedCutoff filter cutoff frequency normalized to Nyquist rate of output + */ + void generateCoefficients(int32_t inputRate, + int32_t outputRate, + int32_t numRows, + double phaseIncrement, + float normalizedCutoff); + + + int32_t getIntegerPhase() { + return mIntegerPhase; + } + + static constexpr int kMaxCoefficients = 8 * 1024; + std::vector mCoefficients; + + const int mNumTaps; + int mCursor = 0; + std::vector mX; // delayed input values for the FIR + std::vector mSingleFrame; // one frame for temporary use + int32_t mIntegerPhase = 0; + int32_t mNumerator = 0; + int32_t mDenominator = 0; + + +private: + +#if MCR_USE_KAISER + KaiserWindow mKaiserWindow; +#else + HyperbolicCosineWindow mCoshWindow; +#endif + + static constexpr float kDefaultNormalizedCutoff = 0.70f; + + const int mChannelCount; +}; + +} +#endif //OBOE_MULTICHANNEL_RESAMPLER_H diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/PolyphaseResampler.cpp b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/PolyphaseResampler.cpp new file mode 100644 index 0000000000..4862ad8a0f --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/PolyphaseResampler.cpp @@ -0,0 +1,61 @@ +/* + * Copyright 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include +#include "IntegerRatio.h" +#include "PolyphaseResampler.h" + +using namespace resampler; + +PolyphaseResampler::PolyphaseResampler(const MultiChannelResampler::Builder &builder) + : MultiChannelResampler(builder) + { + assert((getNumTaps() % 4) == 0); // Required for loop unrolling. + + int32_t inputRate = builder.getInputRate(); + int32_t outputRate = builder.getOutputRate(); + + int32_t numRows = mDenominator; + double phaseIncrement = (double) inputRate / (double) outputRate; + generateCoefficients(inputRate, outputRate, + numRows, phaseIncrement, + builder.getNormalizedCutoff()); +} + +void PolyphaseResampler::readFrame(float *frame) { + // Clear accumulator for mixing. + std::fill(mSingleFrame.begin(), mSingleFrame.end(), 0.0); + +// printf("PolyphaseResampler: mCoefficientCursor = %4d\n", mCoefficientCursor); + // Multiply input times windowed sinc function. + float *coefficients = &mCoefficients[mCoefficientCursor]; + float *xFrame = &mX[mCursor * getChannelCount()]; + for (int i = 0; i < mNumTaps; i++) { + float coefficient = *coefficients++; +// printf("PolyphaseResampler: coeff = %10.6f, xFrame[0] = %10.6f\n", coefficient, xFrame[0]); + for (int channel = 0; channel < getChannelCount(); channel++) { + mSingleFrame[channel] += *xFrame++ * coefficient; + } + } + + // Advance and wrap through coefficients. + mCoefficientCursor = (mCoefficientCursor + mNumTaps) % mCoefficients.size(); + + // Copy accumulator to output. + for (int channel = 0; channel < getChannelCount(); channel++) { + frame[channel] = mSingleFrame[channel]; + } +} diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/PolyphaseResampler.h b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/PolyphaseResampler.h new file mode 100644 index 0000000000..c7de1185c7 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/PolyphaseResampler.h @@ -0,0 +1,51 @@ +/* + * Copyright 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_POLYPHASE_RESAMPLER_H +#define OBOE_POLYPHASE_RESAMPLER_H + +#include +#include +#include +#include +#include "MultiChannelResampler.h" + +namespace resampler { +/** + * Resample that is optimized for a reduced ratio of sample rates. + * All of the coefficients for eacxh possible phase value are precalculated. + */ +class PolyphaseResampler : public MultiChannelResampler { +public: + /** + * + * @param builder containing lots of parameters + */ + explicit PolyphaseResampler(const MultiChannelResampler::Builder &builder); + + virtual ~PolyphaseResampler() = default; + + void readFrame(float *frame) override; + +protected: + + int32_t mCoefficientCursor = 0; + +}; + +} + +#endif //OBOE_POLYPHASE_RESAMPLER_H diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/PolyphaseResamplerMono.cpp b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/PolyphaseResamplerMono.cpp new file mode 100644 index 0000000000..2dcdc8ec33 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/PolyphaseResamplerMono.cpp @@ -0,0 +1,62 @@ +/* + * Copyright 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include "PolyphaseResamplerMono.h" + +using namespace resampler; + +#define MONO 1 + +PolyphaseResamplerMono::PolyphaseResamplerMono(const MultiChannelResampler::Builder &builder) + : PolyphaseResampler(builder) { + assert(builder.getChannelCount() == MONO); +} + +void PolyphaseResamplerMono::writeFrame(const float *frame) { + // Move cursor before write so that cursor points to last written frame in read. + if (--mCursor < 0) { + mCursor = getNumTaps() - 1; + } + float *dest = &mX[mCursor * MONO]; + const int offset = mNumTaps * MONO; + // Write each channel twice so we avoid having to wrap when running the FIR. + const float sample = frame[0]; + // Put ordered writes together. + dest[0] = sample; + dest[offset] = sample; +} + +void PolyphaseResamplerMono::readFrame(float *frame) { + // Clear accumulator. + float sum = 0.0; + + // Multiply input times precomputed windowed sinc function. + const float *coefficients = &mCoefficients[mCoefficientCursor]; + float *xFrame = &mX[mCursor * MONO]; + const int numLoops = mNumTaps >> 2; // n/4 + for (int i = 0; i < numLoops; i++) { + // Manual loop unrolling, might get converted to SIMD. + sum += *xFrame++ * *coefficients++; + sum += *xFrame++ * *coefficients++; + sum += *xFrame++ * *coefficients++; + sum += *xFrame++ * *coefficients++; + } + + mCoefficientCursor = (mCoefficientCursor + mNumTaps) % mCoefficients.size(); + + // Copy accumulator to output. + frame[0] = sum; +} diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/PolyphaseResamplerMono.h b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/PolyphaseResamplerMono.h new file mode 100644 index 0000000000..d97b513cfb --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/PolyphaseResamplerMono.h @@ -0,0 +1,39 @@ +/* + * Copyright 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_POLYPHASE_RESAMPLER_MONO_H +#define OBOE_POLYPHASE_RESAMPLER_MONO_H + +#include +#include +#include "PolyphaseResampler.h" + +namespace resampler { + +class PolyphaseResamplerMono : public PolyphaseResampler { +public: + explicit PolyphaseResamplerMono(const MultiChannelResampler::Builder &builder); + + virtual ~PolyphaseResamplerMono() = default; + + void writeFrame(const float *frame) override; + + void readFrame(float *frame) override; +}; + +} + +#endif //OBOE_POLYPHASE_RESAMPLER_MONO_H diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/PolyphaseResamplerStereo.cpp b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/PolyphaseResamplerStereo.cpp new file mode 100644 index 0000000000..5c73a8e3ba --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/PolyphaseResamplerStereo.cpp @@ -0,0 +1,78 @@ +/* + * Copyright 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include "PolyphaseResamplerStereo.h" + +using namespace resampler; + +#define STEREO 2 + +PolyphaseResamplerStereo::PolyphaseResamplerStereo(const MultiChannelResampler::Builder &builder) + : PolyphaseResampler(builder) { + assert(builder.getChannelCount() == STEREO); +} + +void PolyphaseResamplerStereo::writeFrame(const float *frame) { + // Move cursor before write so that cursor points to last written frame in read. + if (--mCursor < 0) { + mCursor = getNumTaps() - 1; + } + float *dest = &mX[mCursor * STEREO]; + const int offset = mNumTaps * STEREO; + // Write each channel twice so we avoid having to wrap when running the FIR. + const float left = frame[0]; + const float right = frame[1]; + // Put ordered writes together. + dest[0] = left; + dest[1] = right; + dest[offset] = left; + dest[1 + offset] = right; +} + +void PolyphaseResamplerStereo::readFrame(float *frame) { + // Clear accumulators. + float left = 0.0; + float right = 0.0; + + // Multiply input times precomputed windowed sinc function. + const float *coefficients = &mCoefficients[mCoefficientCursor]; + float *xFrame = &mX[mCursor * STEREO]; + const int numLoops = mNumTaps >> 2; // n/4 + for (int i = 0; i < numLoops; i++) { + // Manual loop unrolling, might get converted to SIMD. + float coefficient = *coefficients++; + left += *xFrame++ * coefficient; + right += *xFrame++ * coefficient; + + coefficient = *coefficients++; // next tap + left += *xFrame++ * coefficient; + right += *xFrame++ * coefficient; + + coefficient = *coefficients++; // next tap + left += *xFrame++ * coefficient; + right += *xFrame++ * coefficient; + + coefficient = *coefficients++; // next tap + left += *xFrame++ * coefficient; + right += *xFrame++ * coefficient; + } + + mCoefficientCursor = (mCoefficientCursor + mNumTaps) % mCoefficients.size(); + + // Copy accumulators to output. + frame[0] = left; + frame[1] = right; +} diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/PolyphaseResamplerStereo.h b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/PolyphaseResamplerStereo.h new file mode 100644 index 0000000000..3dfd4e2d3f --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/PolyphaseResamplerStereo.h @@ -0,0 +1,39 @@ +/* + * Copyright 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_POLYPHASE_RESAMPLER_STEREO_H +#define OBOE_POLYPHASE_RESAMPLER_STEREO_H + +#include +#include +#include "PolyphaseResampler.h" + +namespace resampler { + +class PolyphaseResamplerStereo : public PolyphaseResampler { +public: + explicit PolyphaseResamplerStereo(const MultiChannelResampler::Builder &builder); + + virtual ~PolyphaseResamplerStereo() = default; + + void writeFrame(const float *frame) override; + + void readFrame(float *frame) override; +}; + +} + +#endif //OBOE_POLYPHASE_RESAMPLER_STEREO_H diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/SincResampler.cpp b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/SincResampler.cpp new file mode 100644 index 0000000000..1084356dec --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/SincResampler.cpp @@ -0,0 +1,76 @@ +/* + * Copyright 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include +#include "SincResampler.h" + +using namespace resampler; + +SincResampler::SincResampler(const MultiChannelResampler::Builder &builder) + : MultiChannelResampler(builder) + , mSingleFrame2(builder.getChannelCount()) { + assert((getNumTaps() % 4) == 0); // Required for loop unrolling. + mNumRows = kMaxCoefficients / getNumTaps(); // no guard row needed +// printf("SincResampler: numRows = %d\n", mNumRows); + mPhaseScaler = (double) mNumRows / mDenominator; + double phaseIncrement = 1.0 / mNumRows; + generateCoefficients(builder.getInputRate(), + builder.getOutputRate(), + mNumRows, + phaseIncrement, + builder.getNormalizedCutoff()); +} + +void SincResampler::readFrame(float *frame) { + // Clear accumulator for mixing. + std::fill(mSingleFrame.begin(), mSingleFrame.end(), 0.0); + std::fill(mSingleFrame2.begin(), mSingleFrame2.end(), 0.0); + + // Determine indices into coefficients table. + double tablePhase = getIntegerPhase() * mPhaseScaler; + int index1 = static_cast(floor(tablePhase)); + if (index1 >= mNumRows) { // no guard row needed because we wrap the indices + tablePhase -= mNumRows; + index1 -= mNumRows; + } + + int index2 = index1 + 1; + if (index2 >= mNumRows) { // no guard row needed because we wrap the indices + index2 -= mNumRows; + } + + float *coefficients1 = &mCoefficients[index1 * getNumTaps()]; + float *coefficients2 = &mCoefficients[index2 * getNumTaps()]; + + float *xFrame = &mX[mCursor * getChannelCount()]; + for (int i = 0; i < mNumTaps; i++) { + float coefficient1 = *coefficients1++; + float coefficient2 = *coefficients2++; + for (int channel = 0; channel < getChannelCount(); channel++) { + float sample = *xFrame++; + mSingleFrame[channel] += sample * coefficient1; + mSingleFrame2[channel] += sample * coefficient2; + } + } + + // Interpolate and copy to output. + float fraction = tablePhase - index1; + for (int channel = 0; channel < getChannelCount(); channel++) { + float low = mSingleFrame[channel]; + float high = mSingleFrame2[channel]; + frame[channel] = low + (fraction * (high - low)); + } +} diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/SincResampler.h b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/SincResampler.h new file mode 100644 index 0000000000..6ab61c9637 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/SincResampler.h @@ -0,0 +1,47 @@ +/* + * Copyright 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_SINC_RESAMPLER_H +#define OBOE_SINC_RESAMPLER_H + +#include +#include +#include +#include "MultiChannelResampler.h" + +namespace resampler { + +/** + * Resampler that can interpolate between coefficients. + * This can be used to support arbitrary ratios. + */ +class SincResampler : public MultiChannelResampler { +public: + explicit SincResampler(const MultiChannelResampler::Builder &builder); + + virtual ~SincResampler() = default; + + void readFrame(float *frame) override; + +protected: + + std::vector mSingleFrame2; // for interpolation + int32_t mNumRows = 0; + double mPhaseScaler = 1.0; +}; + +} +#endif //OBOE_SINC_RESAMPLER_H diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/SincResamplerStereo.cpp b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/SincResamplerStereo.cpp new file mode 100644 index 0000000000..bde658a486 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/SincResamplerStereo.cpp @@ -0,0 +1,80 @@ +/* + * Copyright 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include + +#include "SincResamplerStereo.h" + +using namespace resampler; + +#define STEREO 2 + +SincResamplerStereo::SincResamplerStereo(const MultiChannelResampler::Builder &builder) + : SincResampler(builder) { + assert(builder.getChannelCount() == STEREO); +} + +void SincResamplerStereo::writeFrame(const float *frame) { + // Move cursor before write so that cursor points to last written frame in read. + if (--mCursor < 0) { + mCursor = getNumTaps() - 1; + } + float *dest = &mX[mCursor * STEREO]; + const int offset = mNumTaps * STEREO; + // Write each channel twice so we avoid having to wrap when running the FIR. + const float left = frame[0]; + const float right = frame[1]; + // Put ordered writes together. + dest[0] = left; + dest[1] = right; + dest[offset] = left; + dest[1 + offset] = right; +} + +// Multiply input times windowed sinc function. +void SincResamplerStereo::readFrame(float *frame) { + // Clear accumulator for mixing. + std::fill(mSingleFrame.begin(), mSingleFrame.end(), 0.0); + std::fill(mSingleFrame2.begin(), mSingleFrame2.end(), 0.0); + + // Determine indices into coefficients table. + double tablePhase = getIntegerPhase() * mPhaseScaler; + int index1 = static_cast(floor(tablePhase)); + float *coefficients1 = &mCoefficients[index1 * getNumTaps()]; + int index2 = (index1 + 1); + if (index2 >= mNumRows) { // no guard row needed because we wrap the indices + index2 = 0; + } + float *coefficients2 = &mCoefficients[index2 * getNumTaps()]; + float *xFrame = &mX[mCursor * getChannelCount()]; + for (int i = 0; i < mNumTaps; i++) { + float coefficient1 = *coefficients1++; + float coefficient2 = *coefficients2++; + for (int channel = 0; channel < getChannelCount(); channel++) { + float sample = *xFrame++; + mSingleFrame[channel] += sample * coefficient1; + mSingleFrame2[channel] += sample * coefficient2; + } + } + + // Interpolate and copy to output. + float fraction = tablePhase - index1; + for (int channel = 0; channel < getChannelCount(); channel++) { + float low = mSingleFrame[channel]; + float high = mSingleFrame2[channel]; + frame[channel] = low + (fraction * (high - low)); + } +} diff --git a/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/SincResamplerStereo.h b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/SincResamplerStereo.h new file mode 100644 index 0000000000..7d66c26fdb --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/flowgraph/resampler/SincResamplerStereo.h @@ -0,0 +1,39 @@ +/* + * Copyright 2019 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_SINC_RESAMPLER_STEREO_H +#define OBOE_SINC_RESAMPLER_STEREO_H + +#include +#include +#include "SincResampler.h" + +namespace resampler { + +class SincResamplerStereo : public SincResampler { +public: + explicit SincResamplerStereo(const MultiChannelResampler::Builder &builder); + + virtual ~SincResamplerStereo() = default; + + void writeFrame(const float *frame) override; + + void readFrame(float *frame) override; + +}; + +} +#endif //OBOE_SINC_RESAMPLER_STEREO_H diff --git a/modules/juce_audio_devices/native/oboe/src/opensles/AudioInputStreamOpenSLES.cpp b/modules/juce_audio_devices/native/oboe/src/opensles/AudioInputStreamOpenSLES.cpp new file mode 100644 index 0000000000..507620264f --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/opensles/AudioInputStreamOpenSLES.cpp @@ -0,0 +1,352 @@ +/* + * Copyright 2017 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include + +#include +#include + +#include "oboe/AudioStreamBuilder.h" +#include "AudioInputStreamOpenSLES.h" +#include "AudioStreamOpenSLES.h" +#include "OpenSLESUtilities.h" + +using namespace oboe; + +static SLuint32 OpenSLES_convertInputPreset(InputPreset oboePreset) { + SLuint32 openslPreset = SL_ANDROID_RECORDING_PRESET_NONE; + switch(oboePreset) { + case InputPreset::Generic: + openslPreset = SL_ANDROID_RECORDING_PRESET_GENERIC; + break; + case InputPreset::Camcorder: + openslPreset = SL_ANDROID_RECORDING_PRESET_CAMCORDER; + break; + case InputPreset::VoiceRecognition: + openslPreset = SL_ANDROID_RECORDING_PRESET_VOICE_RECOGNITION; + break; + case InputPreset::VoiceCommunication: + openslPreset = SL_ANDROID_RECORDING_PRESET_VOICE_COMMUNICATION; + break; + case InputPreset::Unprocessed: + openslPreset = SL_ANDROID_RECORDING_PRESET_UNPROCESSED; + break; + default: + break; + } + return openslPreset; +} + +AudioInputStreamOpenSLES::AudioInputStreamOpenSLES(const AudioStreamBuilder &builder) + : AudioStreamOpenSLES(builder) { +} + +AudioInputStreamOpenSLES::~AudioInputStreamOpenSLES() { +} + +// Calculate masks specific to INPUT streams. +SLuint32 AudioInputStreamOpenSLES::channelCountToChannelMask(int channelCount) const { + // Derived from internal sles_channel_in_mask_from_count(chanCount); + // in "frameworks/wilhelm/src/android/channels.cpp". + // Yes, it seems strange to use SPEAKER constants to describe inputs. + // But that is how OpenSL ES does it internally. + switch (channelCount) { + case 1: + return SL_SPEAKER_FRONT_LEFT; + case 2: + return SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT; + default: + return channelCountToChannelMaskDefault(channelCount); + } +} + +Result AudioInputStreamOpenSLES::open() { + logUnsupportedAttributes(); + + SLAndroidConfigurationItf configItf = nullptr; + + if (getSdkVersion() < __ANDROID_API_M__ && mFormat == AudioFormat::Float){ + // TODO: Allow floating point format on API <23 using float->int16 converter + return Result::ErrorInvalidFormat; + } + + // If audio format is unspecified then choose a suitable default. + // API 23+: FLOAT + // API <23: INT16 + if (mFormat == AudioFormat::Unspecified){ + mFormat = (getSdkVersion() < __ANDROID_API_M__) ? + AudioFormat::I16 : AudioFormat::Float; + } + + Result oboeResult = AudioStreamOpenSLES::open(); + if (Result::OK != oboeResult) return oboeResult; + + SLuint32 bitsPerSample = static_cast(getBytesPerSample() * kBitsPerByte); + + // configure audio sink + SLDataLocator_AndroidSimpleBufferQueue loc_bufq = { + SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE, // locatorType + static_cast(kBufferQueueLength)}; // numBuffers + + // Define the audio data format. + SLDataFormat_PCM format_pcm = { + SL_DATAFORMAT_PCM, // formatType + static_cast(mChannelCount), // numChannels + static_cast(mSampleRate * kMillisPerSecond), // milliSamplesPerSec + bitsPerSample, // bitsPerSample + bitsPerSample, // containerSize; + channelCountToChannelMask(mChannelCount), // channelMask + getDefaultByteOrder(), + }; + + SLDataSink audioSink = {&loc_bufq, &format_pcm}; + + /** + * API 23 (Marshmallow) introduced support for floating-point data representation and an + * extended data format type: SLAndroidDataFormat_PCM_EX for recording streams (playback streams + * got this in API 21). If running on API 23+ use this newer format type, creating it from our + * original format. + */ + SLAndroidDataFormat_PCM_EX format_pcm_ex; + if (getSdkVersion() >= __ANDROID_API_M__) { + SLuint32 representation = OpenSLES_ConvertFormatToRepresentation(getFormat()); + // Fill in the format structure. + format_pcm_ex = OpenSLES_createExtendedFormat(format_pcm, representation); + // Use in place of the previous format. + audioSink.pFormat = &format_pcm_ex; + } + + + // configure audio source + SLDataLocator_IODevice loc_dev = {SL_DATALOCATOR_IODEVICE, + SL_IODEVICE_AUDIOINPUT, + SL_DEFAULTDEVICEID_AUDIOINPUT, + NULL}; + SLDataSource audioSrc = {&loc_dev, NULL}; + + SLresult result = EngineOpenSLES::getInstance().createAudioRecorder(&mObjectInterface, + &audioSrc, + &audioSink); + + if (SL_RESULT_SUCCESS != result) { + LOGE("createAudioRecorder() result:%s", getSLErrStr(result)); + goto error; + } + + // Configure the stream. + result = (*mObjectInterface)->GetInterface(mObjectInterface, + SL_IID_ANDROIDCONFIGURATION, + &configItf); + + if (SL_RESULT_SUCCESS != result) { + LOGW("%s() GetInterface(SL_IID_ANDROIDCONFIGURATION) failed with %s", + __func__, getSLErrStr(result)); + } else { + SLuint32 presetValue = OpenSLES_convertInputPreset(getInputPreset()); + result = (*configItf)->SetConfiguration(configItf, + SL_ANDROID_KEY_RECORDING_PRESET, + &presetValue, + sizeof(SLuint32)); + if (SL_RESULT_SUCCESS != result + && presetValue != SL_ANDROID_RECORDING_PRESET_VOICE_RECOGNITION) { + presetValue = SL_ANDROID_RECORDING_PRESET_VOICE_RECOGNITION; + mInputPreset = InputPreset::VoiceRecognition; + (*configItf)->SetConfiguration(configItf, + SL_ANDROID_KEY_RECORDING_PRESET, + &presetValue, + sizeof(SLuint32)); + } + + result = configurePerformanceMode(configItf); + if (SL_RESULT_SUCCESS != result) { + goto error; + } + } + + result = (*mObjectInterface)->Realize(mObjectInterface, SL_BOOLEAN_FALSE); + if (SL_RESULT_SUCCESS != result) { + LOGE("Realize recorder object result:%s", getSLErrStr(result)); + goto error; + } + + result = (*mObjectInterface)->GetInterface(mObjectInterface, SL_IID_RECORD, &mRecordInterface); + if (SL_RESULT_SUCCESS != result) { + LOGE("GetInterface RECORD result:%s", getSLErrStr(result)); + goto error; + } + + result = AudioStreamOpenSLES::registerBufferQueueCallback(); + if (SL_RESULT_SUCCESS != result) { + goto error; + } + + result = updateStreamParameters(configItf); + if (SL_RESULT_SUCCESS != result) { + goto error; + } + + oboeResult = configureBufferSizes(); + if (Result::OK != oboeResult) { + goto error; + } + + allocateFifo(); + + setState(StreamState::Open); + return Result::OK; + +error: + return Result::ErrorInternal; // TODO convert error from SLES to OBOE +} + +Result AudioInputStreamOpenSLES::close() { + LOGD("AudioInputStreamOpenSLES::%s()", __func__); + mLock.lock(); + Result result = Result::OK; + if (getState() == StreamState::Closed){ + result = Result::ErrorClosed; + } else { + mLock.unlock(); // avoid recursive lock + requestStop(); + mLock.lock(); + // invalidate any interfaces + mRecordInterface = nullptr; + result = AudioStreamOpenSLES::close(); + } + mLock.unlock(); // avoid recursive lock + return result; +} + +Result AudioInputStreamOpenSLES::setRecordState_l(SLuint32 newState) { + LOGD("AudioInputStreamOpenSLES::%s(%u)", __func__, newState); + Result result = Result::OK; + + if (mRecordInterface == nullptr) { + LOGE("AudioInputStreamOpenSLES::%s() mRecordInterface is null", __func__); + return Result::ErrorInvalidState; + } + SLresult slResult = (*mRecordInterface)->SetRecordState(mRecordInterface, newState); + //LOGD("AudioInputStreamOpenSLES::%s(%u) returned %u", __func__, newState, slResult); + if (SL_RESULT_SUCCESS != slResult) { + LOGE("AudioInputStreamOpenSLES::%s(%u) returned error %s", + __func__, newState, getSLErrStr(slResult)); + result = Result::ErrorInternal; // TODO review + } + return result; +} + +Result AudioInputStreamOpenSLES::requestStart() { + LOGD("AudioInputStreamOpenSLES(): %s() called", __func__); + std::lock_guard lock(mLock); + StreamState initialState = getState(); + switch (initialState) { + case StreamState::Starting: + case StreamState::Started: + return Result::OK; + case StreamState::Closed: + return Result::ErrorClosed; + default: + break; + } + + // We use a callback if the user requests one + // OR if we have an internal callback to fill the blocking IO buffer. + setDataCallbackEnabled(true); + + setState(StreamState::Starting); + Result result = setRecordState_l(SL_RECORDSTATE_RECORDING); + if (result == Result::OK) { + setState(StreamState::Started); + // Enqueue the first buffer to start the streaming. + // This does not call the callback function. + enqueueCallbackBuffer(mSimpleBufferQueueInterface); + } else { + setState(initialState); + } + return result; +} + + +Result AudioInputStreamOpenSLES::requestPause() { + LOGW("AudioInputStreamOpenSLES::%s() is intentionally not implemented for input " + "streams", __func__); + return Result::ErrorUnimplemented; // Matches AAudio behavior. +} + +Result AudioInputStreamOpenSLES::requestFlush() { + LOGW("AudioInputStreamOpenSLES::%s() is intentionally not implemented for input " + "streams", __func__); + return Result::ErrorUnimplemented; // Matches AAudio behavior. +} + +Result AudioInputStreamOpenSLES::requestStop() { + LOGD("AudioInputStreamOpenSLES(): %s() called", __func__); + + std::lock_guard lock(mLock); + StreamState initialState = getState(); + switch (initialState) { + case StreamState::Stopping: + case StreamState::Stopped: + return Result::OK; + case StreamState::Closed: + return Result::ErrorClosed; + default: + break; + } + + setState(StreamState::Stopping); + + Result result = setRecordState_l(SL_RECORDSTATE_STOPPED); + if (result == Result::OK) { + mPositionMillis.reset32(); // OpenSL ES resets its millisecond position when stopped. + setState(StreamState::Stopped); + } else { + setState(initialState); + } + return result; +} + +void AudioInputStreamOpenSLES::updateFramesWritten() { + if (usingFIFO()) { + AudioStreamBuffered::updateFramesWritten(); + } else { + mFramesWritten = getFramesProcessedByServer(); + } +} + +Result AudioInputStreamOpenSLES::updateServiceFrameCounter() { + Result result = Result::OK; + // Avoid deadlock if another thread is trying to stop or close this stream + // and this is being called from a callback. + if (mLock.try_lock()) { + + if (mRecordInterface == nullptr) { + mLock.unlock(); + return Result::ErrorNull; + } + SLmillisecond msec = 0; + SLresult slResult = (*mRecordInterface)->GetPosition(mRecordInterface, &msec); + if (SL_RESULT_SUCCESS != slResult) { + LOGD("%s(): GetPosition() returned %s", __func__, getSLErrStr(slResult)); + // set result based on SLresult + result = Result::ErrorInternal; + } else { + mPositionMillis.update32(msec); + } + mLock.unlock(); + } + return result; +} diff --git a/modules/juce_audio_devices/native/oboe/src/opensles/AudioInputStreamOpenSLES.h b/modules/juce_audio_devices/native/oboe/src/opensles/AudioInputStreamOpenSLES.h new file mode 100644 index 0000000000..7be1c96d77 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/opensles/AudioInputStreamOpenSLES.h @@ -0,0 +1,65 @@ +/* + * Copyright 2017 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef AUDIO_INPUT_STREAM_OPENSL_ES_H_ +#define AUDIO_INPUT_STREAM_OPENSL_ES_H_ + + +#include +#include + +#include "oboe/Oboe.h" +#include "AudioStreamOpenSLES.h" + +namespace oboe { + +/** + * INTERNAL USE ONLY + */ + +class AudioInputStreamOpenSLES : public AudioStreamOpenSLES { +public: + AudioInputStreamOpenSLES(); + explicit AudioInputStreamOpenSLES(const AudioStreamBuilder &builder); + + virtual ~AudioInputStreamOpenSLES(); + + Result open() override; + Result close() override; + + Result requestStart() override; + Result requestPause() override; + Result requestFlush() override; + Result requestStop() override; + +protected: + + Result updateServiceFrameCounter() override; + + void updateFramesWritten() override; + +private: + + SLuint32 channelCountToChannelMask(int chanCount) const; + + Result setRecordState_l(SLuint32 newState); + + SLRecordItf mRecordInterface = nullptr; +}; + +} // namespace oboe + +#endif //AUDIO_INPUT_STREAM_OPENSL_ES_H_ diff --git a/modules/juce_audio_devices/native/oboe/src/opensles/AudioOutputStreamOpenSLES.cpp b/modules/juce_audio_devices/native/oboe/src/opensles/AudioOutputStreamOpenSLES.cpp new file mode 100644 index 0000000000..a8c70b607b --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/opensles/AudioOutputStreamOpenSLES.cpp @@ -0,0 +1,452 @@ +/* + * Copyright 2017 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include + +#include +#include +#include + +#include "oboe/AudioStreamBuilder.h" +#include "AudioOutputStreamOpenSLES.h" +#include "AudioStreamOpenSLES.h" +#include "OpenSLESUtilities.h" +#include "OutputMixerOpenSLES.h" + +using namespace oboe; + +static SLuint32 OpenSLES_convertOutputUsage(Usage oboeUsage) { + SLuint32 openslStream = SL_ANDROID_STREAM_MEDIA; + switch(oboeUsage) { + case Usage::Media: + openslStream = SL_ANDROID_STREAM_MEDIA; + break; + case Usage::VoiceCommunication: + case Usage::VoiceCommunicationSignalling: + openslStream = SL_ANDROID_STREAM_VOICE; + break; + case Usage::Alarm: + openslStream = SL_ANDROID_STREAM_ALARM; + break; + case Usage::Notification: + case Usage::NotificationRingtone: + case Usage::NotificationEvent: + openslStream = SL_ANDROID_STREAM_NOTIFICATION; + break; + case Usage::AssistanceAccessibility: + case Usage::AssistanceNavigationGuidance: + case Usage::AssistanceSonification: + openslStream = SL_ANDROID_STREAM_SYSTEM; + break; + case Usage::Game: + openslStream = SL_ANDROID_STREAM_MEDIA; + break; + case Usage::Assistant: + default: + openslStream = SL_ANDROID_STREAM_SYSTEM; + break; + } + return openslStream; +} + +AudioOutputStreamOpenSLES::AudioOutputStreamOpenSLES(const AudioStreamBuilder &builder) + : AudioStreamOpenSLES(builder) { +} + +// These will wind up in +constexpr int SL_ANDROID_SPEAKER_STEREO = (SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT); + +constexpr int SL_ANDROID_SPEAKER_QUAD = (SL_ANDROID_SPEAKER_STEREO + | SL_SPEAKER_BACK_LEFT | SL_SPEAKER_BACK_RIGHT); + +constexpr int SL_ANDROID_SPEAKER_5DOT1 = (SL_ANDROID_SPEAKER_QUAD + | SL_SPEAKER_FRONT_CENTER | SL_SPEAKER_LOW_FREQUENCY); + +constexpr int SL_ANDROID_SPEAKER_7DOT1 = (SL_ANDROID_SPEAKER_5DOT1 | SL_SPEAKER_SIDE_LEFT + | SL_SPEAKER_SIDE_RIGHT); + +SLuint32 AudioOutputStreamOpenSLES::channelCountToChannelMask(int channelCount) const { + SLuint32 channelMask = 0; + + switch (channelCount) { + case 1: + channelMask = SL_SPEAKER_FRONT_CENTER; + break; + + case 2: + channelMask = SL_ANDROID_SPEAKER_STEREO; + break; + + case 4: // Quad + channelMask = SL_ANDROID_SPEAKER_QUAD; + break; + + case 6: // 5.1 + channelMask = SL_ANDROID_SPEAKER_5DOT1; + break; + + case 8: // 7.1 + channelMask = SL_ANDROID_SPEAKER_7DOT1; + break; + + default: + channelMask = channelCountToChannelMaskDefault(channelCount); + break; + } + return channelMask; +} + +Result AudioOutputStreamOpenSLES::open() { + logUnsupportedAttributes(); + + SLAndroidConfigurationItf configItf = nullptr; + + + if (getSdkVersion() < __ANDROID_API_L__ && mFormat == AudioFormat::Float){ + // TODO: Allow floating point format on API <21 using float->int16 converter + return Result::ErrorInvalidFormat; + } + + // If audio format is unspecified then choose a suitable default. + // API 21+: FLOAT + // API <21: INT16 + if (mFormat == AudioFormat::Unspecified){ + mFormat = (getSdkVersion() < __ANDROID_API_L__) ? + AudioFormat::I16 : AudioFormat::Float; + } + + Result oboeResult = AudioStreamOpenSLES::open(); + if (Result::OK != oboeResult) return oboeResult; + + SLresult result = OutputMixerOpenSL::getInstance().open(); + if (SL_RESULT_SUCCESS != result) { + AudioStreamOpenSLES::close(); + return Result::ErrorInternal; + } + + SLuint32 bitsPerSample = static_cast(getBytesPerSample() * kBitsPerByte); + + // configure audio source + SLDataLocator_AndroidSimpleBufferQueue loc_bufq = { + SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE, // locatorType + static_cast(kBufferQueueLength)}; // numBuffers + + // Define the audio data format. + SLDataFormat_PCM format_pcm = { + SL_DATAFORMAT_PCM, // formatType + static_cast(mChannelCount), // numChannels + static_cast(mSampleRate * kMillisPerSecond), // milliSamplesPerSec + bitsPerSample, // bitsPerSample + bitsPerSample, // containerSize; + channelCountToChannelMask(mChannelCount), // channelMask + getDefaultByteOrder(), + }; + + SLDataSource audioSrc = {&loc_bufq, &format_pcm}; + + /** + * API 21 (Lollipop) introduced support for floating-point data representation and an extended + * data format type: SLAndroidDataFormat_PCM_EX. If running on API 21+ use this newer format + * type, creating it from our original format. + */ + SLAndroidDataFormat_PCM_EX format_pcm_ex; + if (getSdkVersion() >= __ANDROID_API_L__) { + SLuint32 representation = OpenSLES_ConvertFormatToRepresentation(getFormat()); + // Fill in the format structure. + format_pcm_ex = OpenSLES_createExtendedFormat(format_pcm, representation); + // Use in place of the previous format. + audioSrc.pFormat = &format_pcm_ex; + } + + result = OutputMixerOpenSL::getInstance().createAudioPlayer(&mObjectInterface, + &audioSrc); + if (SL_RESULT_SUCCESS != result) { + LOGE("createAudioPlayer() result:%s", getSLErrStr(result)); + goto error; + } + + // Configure the stream. + result = (*mObjectInterface)->GetInterface(mObjectInterface, + SL_IID_ANDROIDCONFIGURATION, + (void *)&configItf); + if (SL_RESULT_SUCCESS != result) { + LOGW("%s() GetInterface(SL_IID_ANDROIDCONFIGURATION) failed with %s", + __func__, getSLErrStr(result)); + } else { + result = configurePerformanceMode(configItf); + if (SL_RESULT_SUCCESS != result) { + goto error; + } + + SLuint32 presetValue = OpenSLES_convertOutputUsage(getUsage()); + result = (*configItf)->SetConfiguration(configItf, + SL_ANDROID_KEY_STREAM_TYPE, + &presetValue, + sizeof(presetValue)); + if (SL_RESULT_SUCCESS != result) { + goto error; + } + } + + result = (*mObjectInterface)->Realize(mObjectInterface, SL_BOOLEAN_FALSE); + if (SL_RESULT_SUCCESS != result) { + LOGE("Realize player object result:%s", getSLErrStr(result)); + goto error; + } + + result = (*mObjectInterface)->GetInterface(mObjectInterface, SL_IID_PLAY, &mPlayInterface); + if (SL_RESULT_SUCCESS != result) { + LOGE("GetInterface PLAY result:%s", getSLErrStr(result)); + goto error; + } + + result = AudioStreamOpenSLES::registerBufferQueueCallback(); + if (SL_RESULT_SUCCESS != result) { + goto error; + } + + result = updateStreamParameters(configItf); + if (SL_RESULT_SUCCESS != result) { + goto error; + } + + oboeResult = configureBufferSizes(); + if (Result::OK != oboeResult) { + goto error; + } + + allocateFifo(); + + setState(StreamState::Open); + return Result::OK; + +error: + return Result::ErrorInternal; // TODO convert error from SLES to OBOE +} + +Result AudioOutputStreamOpenSLES::onAfterDestroy() { + OutputMixerOpenSL::getInstance().close(); + return Result::OK; +} + +Result AudioOutputStreamOpenSLES::close() { + mLock.lock(); + Result result = Result::OK; + if (getState() == StreamState::Closed){ + result = Result::ErrorClosed; + } else { + mLock.unlock(); // avoid recursive lock + requestPause(); + mLock.lock(); + // invalidate any interfaces + mPlayInterface = nullptr; + result = AudioStreamOpenSLES::close(); + } + mLock.unlock(); // avoid recursive lock + return result; +} + +Result AudioOutputStreamOpenSLES::setPlayState_l(SLuint32 newState) { + + LOGD("AudioOutputStreamOpenSLES(): %s() called", __func__); + Result result = Result::OK; + + if (mPlayInterface == nullptr){ + LOGE("AudioOutputStreamOpenSLES::%s() mPlayInterface is null", __func__); + return Result::ErrorInvalidState; + } + + SLresult slResult = (*mPlayInterface)->SetPlayState(mPlayInterface, newState); + if (SL_RESULT_SUCCESS != slResult) { + LOGD("AudioOutputStreamOpenSLES(): %s() returned %s", __func__, getSLErrStr(slResult)); + result = Result::ErrorInternal; // TODO convert slResult to Result::Error + } + return result; +} + +Result AudioOutputStreamOpenSLES::requestStart() { + LOGD("AudioOutputStreamOpenSLES(): %s() called", __func__); + + mLock.lock(); + StreamState initialState = getState(); + switch (initialState) { + case StreamState::Starting: + case StreamState::Started: + mLock.unlock(); + return Result::OK; + case StreamState::Closed: + mLock.unlock(); + return Result::ErrorClosed; + default: + break; + } + + // We use a callback if the user requests one + // OR if we have an internal callback to read the blocking IO buffer. + setDataCallbackEnabled(true); + + setState(StreamState::Starting); + Result result = setPlayState_l(SL_PLAYSTATE_PLAYING); + if (result == Result::OK) { + setState(StreamState::Started); + mLock.unlock(); + if (getBufferDepth(mSimpleBufferQueueInterface) == 0) { + // Enqueue the first buffer if needed to start the streaming. + // This might call requestStop() so try to avoid a recursive lock. + processBufferCallback(mSimpleBufferQueueInterface); + } + } else { + setState(initialState); + mLock.unlock(); + } + LOGD("AudioOutputStreamOpenSLES(): %s() returning %d", __func__, result); + return result; +} + +Result AudioOutputStreamOpenSLES::requestPause() { + LOGD("AudioOutputStreamOpenSLES(): %s() called", __func__); + + std::lock_guard lock(mLock); + StreamState initialState = getState(); + switch (initialState) { + case StreamState::Pausing: + case StreamState::Paused: + return Result::OK; + case StreamState::Closed: + return Result::ErrorClosed; + default: + break; + } + + setState(StreamState::Pausing); + Result result = setPlayState_l(SL_PLAYSTATE_PAUSED); + if (result == Result::OK) { + // Note that OpenSL ES does NOT reset its millisecond position when OUTPUT is paused. + int64_t framesWritten = getFramesWritten(); + if (framesWritten >= 0) { + setFramesRead(framesWritten); + } + setState(StreamState::Paused); + } else { + setState(initialState); + } + LOGD("AudioOutputStreamOpenSLES(): %s() returning %d", __func__, result); + return result; +} + +/** + * Flush/clear the queue buffers + */ +Result AudioOutputStreamOpenSLES::requestFlush() { + std::lock_guard lock(mLock); + return requestFlush_l(); +} + +Result AudioOutputStreamOpenSLES::requestFlush_l() { + LOGD("AudioOutputStreamOpenSLES(): %s() called", __func__); + if (getState() == StreamState::Closed) { + return Result::ErrorClosed; + } + + Result result = Result::OK; + if (mPlayInterface == nullptr || mSimpleBufferQueueInterface == nullptr) { + result = Result::ErrorInvalidState; + } else { + SLresult slResult = (*mSimpleBufferQueueInterface)->Clear(mSimpleBufferQueueInterface); + if (slResult != SL_RESULT_SUCCESS){ + LOGW("Failed to clear buffer queue. OpenSLES error: %d", result); + result = Result::ErrorInternal; + } + } + LOGD("AudioOutputStreamOpenSLES(): %s() returning %d", __func__, result); + return result; +} + +Result AudioOutputStreamOpenSLES::requestStop() { + LOGD("AudioOutputStreamOpenSLES(): %s() called", __func__); + + std::lock_guard lock(mLock); + StreamState initialState = getState(); + switch (initialState) { + case StreamState::Stopping: + case StreamState::Stopped: + return Result::OK; + case StreamState::Closed: + return Result::ErrorClosed; + default: + break; + } + + setState(StreamState::Stopping); + + Result result = setPlayState_l(SL_PLAYSTATE_STOPPED); + if (result == Result::OK) { + + // Also clear the buffer queue so the old data won't be played if the stream is restarted. + // Call the _l function that expects to already be under a lock. + if (requestFlush_l() != Result::OK) { + LOGW("Failed to flush the stream. Error %s", convertToText(flush())); + } + + mPositionMillis.reset32(); // OpenSL ES resets its millisecond position when stopped. + int64_t framesWritten = getFramesWritten(); + if (framesWritten >= 0) { + setFramesRead(framesWritten); + } + setState(StreamState::Stopped); + } else { + setState(initialState); + } + LOGD("AudioOutputStreamOpenSLES(): %s() returning %d", __func__, result); + return result; +} + +void AudioOutputStreamOpenSLES::setFramesRead(int64_t framesRead) { + int64_t millisWritten = framesRead * kMillisPerSecond / getSampleRate(); + mPositionMillis.set(millisWritten); +} + +void AudioOutputStreamOpenSLES::updateFramesRead() { + if (usingFIFO()) { + AudioStreamBuffered::updateFramesRead(); + } else { + mFramesRead = getFramesProcessedByServer(); + } +} + +Result AudioOutputStreamOpenSLES::updateServiceFrameCounter() { + Result result = Result::OK; + // Avoid deadlock if another thread is trying to stop or close this stream + // and this is being called from a callback. + if (mLock.try_lock()) { + + if (mPlayInterface == nullptr) { + mLock.unlock(); + return Result::ErrorNull; + } + SLmillisecond msec = 0; + SLresult slResult = (*mPlayInterface)->GetPosition(mPlayInterface, &msec); + if (SL_RESULT_SUCCESS != slResult) { + LOGD("%s(): GetPosition() returned %s", __func__, getSLErrStr(slResult)); + // set result based on SLresult + result = Result::ErrorInternal; + } else { + mPositionMillis.update32(msec); + } + mLock.unlock(); + } + return result; +} diff --git a/modules/juce_audio_devices/native/oboe/src/opensles/AudioOutputStreamOpenSLES.h b/modules/juce_audio_devices/native/oboe/src/opensles/AudioOutputStreamOpenSLES.h new file mode 100644 index 0000000000..d74faf67cf --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/opensles/AudioOutputStreamOpenSLES.h @@ -0,0 +1,77 @@ +/* + * Copyright 2017 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef AUDIO_OUTPUT_STREAM_OPENSL_ES_H_ +#define AUDIO_OUTPUT_STREAM_OPENSL_ES_H_ + + +#include +#include + +#include "oboe/Oboe.h" +#include "AudioStreamOpenSLES.h" + +namespace oboe { + +/** + * INTERNAL USE ONLY + */ +class AudioOutputStreamOpenSLES : public AudioStreamOpenSLES { +public: + AudioOutputStreamOpenSLES(); + explicit AudioOutputStreamOpenSLES(const AudioStreamBuilder &builder); + + virtual ~AudioOutputStreamOpenSLES() = default; + + Result open() override; + Result close() override; + + Result requestStart() override; + Result requestPause() override; + Result requestFlush() override; + Result requestStop() override; + +protected: + + void setFramesRead(int64_t framesRead); + + Result updateServiceFrameCounter() override; + + void updateFramesRead() override; + +private: + + SLuint32 channelCountToChannelMask(int chanCount) const; + + Result onAfterDestroy() override; + + Result requestFlush_l(); + + /** + * Set OpenSL ES PLAYSTATE. + * + * @param newState SL_PLAYSTATE_PAUSED, SL_PLAYSTATE_PLAYING, SL_PLAYSTATE_STOPPED + * @return + */ + Result setPlayState_l(SLuint32 newState); + + SLPlayItf mPlayInterface = nullptr; + +}; + +} // namespace oboe + +#endif //AUDIO_OUTPUT_STREAM_OPENSL_ES_H_ diff --git a/modules/juce_audio_devices/native/oboe/src/opensles/AudioStreamBuffered.cpp b/modules/juce_audio_devices/native/oboe/src/opensles/AudioStreamBuffered.cpp new file mode 100644 index 0000000000..85b9f8e4a5 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/opensles/AudioStreamBuffered.cpp @@ -0,0 +1,266 @@ +/* + * Copyright (C) 2016 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include + +#include "oboe/Oboe.h" + +#include "opensles/AudioStreamBuffered.h" +#include "common/AudioClock.h" + +namespace oboe { + +constexpr int kDefaultBurstsPerBuffer = 16; // arbitrary, allows dynamic latency tuning +constexpr int kMinBurstsPerBuffer = 4; // arbitrary, allows dynamic latency tuning +constexpr int kMinFramesPerBuffer = 48 * 32; // arbitrary + +/* + * AudioStream with a FifoBuffer + */ +AudioStreamBuffered::AudioStreamBuffered(const AudioStreamBuilder &builder) + : AudioStream(builder) { +} + +void AudioStreamBuffered::allocateFifo() { + // If the caller does not provide a callback use our own internal + // callback that reads data from the FIFO. + if (usingFIFO()) { + // FIFO is configured with the same format and channels as the stream. + int32_t capacityFrames = getBufferCapacityInFrames(); + if (capacityFrames == oboe::kUnspecified) { + capacityFrames = getFramesPerBurst() * kDefaultBurstsPerBuffer; + } else { + int32_t minFramesPerBufferByBursts = getFramesPerBurst() * kMinBurstsPerBuffer; + if (capacityFrames <= minFramesPerBufferByBursts) { + capacityFrames = minFramesPerBufferByBursts; + } else { + capacityFrames = std::max(kMinFramesPerBuffer, capacityFrames); + // round up to nearest burst + int32_t numBursts = (capacityFrames + getFramesPerBurst() - 1) + / getFramesPerBurst(); + capacityFrames = numBursts * getFramesPerBurst(); + } + } + // TODO consider using std::make_unique if we require c++14 + mFifoBuffer.reset(new FifoBuffer(getBytesPerFrame(), capacityFrames)); + mBufferCapacityInFrames = capacityFrames; + } +} + +void AudioStreamBuffered::updateFramesWritten() { + if (mFifoBuffer) { + mFramesWritten = static_cast(mFifoBuffer->getWriteCounter()); + } // or else it will get updated by processBufferCallback() +} + +void AudioStreamBuffered::updateFramesRead() { + if (mFifoBuffer) { + mFramesRead = static_cast(mFifoBuffer->getReadCounter()); + } // or else it will get updated by processBufferCallback() +} + +// This is called by the OpenSL ES callback to read or write the back end of the FIFO. +DataCallbackResult AudioStreamBuffered::onDefaultCallback(void *audioData, int numFrames) { + int32_t framesTransferred = 0; + + if (getDirection() == oboe::Direction::Output) { + // Read from the FIFO and write to audioData, clear part of buffer if not enough data. + framesTransferred = mFifoBuffer->readNow(audioData, numFrames); + } else { + // Read from audioData and write to the FIFO + framesTransferred = mFifoBuffer->write(audioData, numFrames); // There is no writeNow() + } + + if (framesTransferred < numFrames) { + LOGD("AudioStreamBuffered::%s(): xrun! framesTransferred = %d, numFrames = %d", + __func__, framesTransferred, numFrames); + // TODO If we do not allow FIFO to wrap then our timestamps will drift when there is an XRun! + incrementXRunCount(); + } + markCallbackTime(static_cast(numFrames)); // so foreground knows how long to wait. + return DataCallbackResult::Continue; +} + +void AudioStreamBuffered::markCallbackTime(int32_t numFrames) { + mLastBackgroundSize = numFrames; + mBackgroundRanAtNanoseconds = AudioClock::getNanoseconds(); +} + +int64_t AudioStreamBuffered::predictNextCallbackTime() { + if (mBackgroundRanAtNanoseconds == 0) { + return 0; + } + int64_t nanosPerBuffer = (kNanosPerSecond * mLastBackgroundSize) / getSampleRate(); + const int64_t margin = 200 * kNanosPerMicrosecond; // arbitrary delay so we wake up just after + return mBackgroundRanAtNanoseconds + nanosPerBuffer + margin; +} + +// Common code for read/write. +// @return Result::OK with frames read/written, or Result::Error* +ResultWithValue AudioStreamBuffered::transfer(void *buffer, + int32_t numFrames, + int64_t timeoutNanoseconds) { + // Validate arguments. + if (buffer == nullptr) { + LOGE("AudioStreamBuffered::%s(): buffer is NULL", __func__); + return ResultWithValue(Result::ErrorNull); + } + if (numFrames < 0) { + LOGE("AudioStreamBuffered::%s(): numFrames is negative", __func__); + return ResultWithValue(Result::ErrorOutOfRange); + } else if (numFrames == 0) { + return ResultWithValue(numFrames); + } + if (timeoutNanoseconds < 0) { + LOGE("AudioStreamBuffered::%s(): timeoutNanoseconds is negative", __func__); + return ResultWithValue(Result::ErrorOutOfRange); + } + + int32_t result = 0; + uint8_t *data = reinterpret_cast(buffer); + int32_t framesLeft = numFrames; + int64_t timeToQuit = 0; + bool repeat = true; + + // Calculate when to timeout. + if (timeoutNanoseconds > 0) { + timeToQuit = AudioClock::getNanoseconds() + timeoutNanoseconds; + } + + // Loop until we get the data, or we have an error, or we timeout. + do { + // read or write + if (getDirection() == Direction::Input) { + result = mFifoBuffer->read(data, framesLeft); + } else { + // between zero and capacity + uint32_t fullFrames = mFifoBuffer->getFullFramesAvailable(); + // Do not write above threshold size. + int32_t emptyFrames = getBufferSizeInFrames() - static_cast(fullFrames); + int32_t framesToWrite = std::max(0, std::min(framesLeft, emptyFrames)); + result = mFifoBuffer->write(data, framesToWrite); + } + if (result > 0) { + data += mFifoBuffer->convertFramesToBytes(result); + framesLeft -= result; + } + + // If we need more data then sleep and try again. + if (framesLeft > 0 && result >= 0 && timeoutNanoseconds > 0) { + int64_t timeNow = AudioClock::getNanoseconds(); + if (timeNow >= timeToQuit) { + LOGE("AudioStreamBuffered::%s(): TIMEOUT", __func__); + repeat = false; // TIMEOUT + } else { + // Figure out how long to sleep. + int64_t sleepForNanos; + int64_t wakeTimeNanos = predictNextCallbackTime(); + if (wakeTimeNanos <= 0) { + // No estimate available. Sleep for one burst. + sleepForNanos = (getFramesPerBurst() * kNanosPerSecond) / getSampleRate(); + } else { + // Don't sleep past timeout. + if (wakeTimeNanos > timeToQuit) { + wakeTimeNanos = timeToQuit; + } + sleepForNanos = wakeTimeNanos - timeNow; + // Avoid rapid loop with no sleep. + const int64_t minSleepTime = kNanosPerMillisecond; // arbitrary + if (sleepForNanos < minSleepTime) { + sleepForNanos = minSleepTime; + } + } + + AudioClock::sleepForNanos(sleepForNanos); + } + + } else { + repeat = false; + } + } while(repeat); + + if (result < 0) { + return ResultWithValue(static_cast(result)); + } else { + int32_t framesWritten = numFrames - framesLeft; + return ResultWithValue(framesWritten); + } +} + +// Write to the FIFO so the callback can read from it. +ResultWithValue AudioStreamBuffered::write(const void *buffer, + int32_t numFrames, + int64_t timeoutNanoseconds) { + if (getState() == StreamState::Closed){ + return ResultWithValue(Result::ErrorClosed); + } + + if (getDirection() == Direction::Input) { + return ResultWithValue(Result::ErrorUnavailable); // TODO review, better error code? + } + updateServiceFrameCounter(); + return transfer(const_cast(buffer), numFrames, timeoutNanoseconds); +} + +// Read data from the FIFO that was written by the callback. +ResultWithValue AudioStreamBuffered::read(void *buffer, + int32_t numFrames, + int64_t timeoutNanoseconds) { + if (getState() == StreamState::Closed){ + return ResultWithValue(Result::ErrorClosed); + } + + if (getDirection() == Direction::Output) { + return ResultWithValue(Result::ErrorUnavailable); // TODO review, better error code? + } + updateServiceFrameCounter(); + return transfer(buffer, numFrames, timeoutNanoseconds); +} + +// Only supported when we are not using a callback. +ResultWithValue AudioStreamBuffered::setBufferSizeInFrames(int32_t requestedFrames) +{ + if (getState() == StreamState::Closed){ + return ResultWithValue(Result::ErrorClosed); + } + + if (!mFifoBuffer) { + return ResultWithValue(Result::ErrorUnimplemented); + } + + if (requestedFrames > mFifoBuffer->getBufferCapacityInFrames()) { + requestedFrames = mFifoBuffer->getBufferCapacityInFrames(); + } else if (requestedFrames < getFramesPerBurst()) { + requestedFrames = getFramesPerBurst(); + } + mBufferSizeInFrames = requestedFrames; + return ResultWithValue(requestedFrames); +} + +int32_t AudioStreamBuffered::getBufferCapacityInFrames() const { + if (mFifoBuffer) { + return mFifoBuffer->getBufferCapacityInFrames(); + } else { + return AudioStream::getBufferCapacityInFrames(); + } +} + +bool AudioStreamBuffered::isXRunCountSupported() const { + // XRun count is only supported if we're using blocking I/O (not callbacks) + return (getCallback() == nullptr); +} + +} // namespace oboe \ No newline at end of file diff --git a/modules/juce_audio_devices/native/oboe/src/opensles/AudioStreamBuffered.h b/modules/juce_audio_devices/native/oboe/src/opensles/AudioStreamBuffered.h new file mode 100644 index 0000000000..5923e8db3a --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/opensles/AudioStreamBuffered.h @@ -0,0 +1,92 @@ +/* + * Copyright (C) 2016 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_STREAM_BUFFERED_H +#define OBOE_STREAM_BUFFERED_H + +#include +#include +#include "common/OboeDebug.h" +#include "oboe/AudioStream.h" +#include "oboe/AudioStreamCallback.h" +#include "fifo/FifoBuffer.h" + +namespace oboe { + +// A stream that contains a FIFO buffer. +// This is used to implement blocking reads and writes. +class AudioStreamBuffered : public AudioStream { +public: + + AudioStreamBuffered(); + explicit AudioStreamBuffered(const AudioStreamBuilder &builder); + + void allocateFifo(); + + + ResultWithValue write(const void *buffer, + int32_t numFrames, + int64_t timeoutNanoseconds) override; + + ResultWithValue read(void *buffer, + int32_t numFrames, + int64_t timeoutNanoseconds) override; + + ResultWithValue setBufferSizeInFrames(int32_t requestedFrames) override; + + int32_t getBufferCapacityInFrames() const override; + + ResultWithValue getXRunCount() const override { + return ResultWithValue(mXRunCount); + } + + bool isXRunCountSupported() const override; + +protected: + + DataCallbackResult onDefaultCallback(void *audioData, int numFrames) override; + + // If there is no callback then we need a FIFO between the App and OpenSL ES. + bool usingFIFO() const { return getCallback() == nullptr; } + + virtual Result updateServiceFrameCounter() = 0; + + void updateFramesRead() override; + void updateFramesWritten() override; + +private: + + int64_t predictNextCallbackTime(); + + void markCallbackTime(int32_t numFrames); + + // Read or write to the FIFO. + ResultWithValue transfer(void *buffer, int32_t numFrames, int64_t timeoutNanoseconds); + + void incrementXRunCount() { + ++mXRunCount; + } + + std::unique_ptr mFifoBuffer{}; + + int64_t mBackgroundRanAtNanoseconds = 0; + int32_t mLastBackgroundSize = 0; + int32_t mXRunCount = 0; +}; + +} // namespace oboe + +#endif //OBOE_STREAM_BUFFERED_H diff --git a/modules/juce_audio_devices/native/oboe/src/opensles/AudioStreamOpenSLES.cpp b/modules/juce_audio_devices/native/oboe/src/opensles/AudioStreamOpenSLES.cpp new file mode 100644 index 0000000000..38bdf1a3d3 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/opensles/AudioStreamOpenSLES.cpp @@ -0,0 +1,387 @@ +/* Copyright 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ +#include +#include +#include + + +#include +#include +#include +#include + +#include "common/OboeDebug.h" +#include "oboe/AudioStreamBuilder.h" +#include "AudioStreamOpenSLES.h" +#include "OpenSLESUtilities.h" + +using namespace oboe; + +AudioStreamOpenSLES::AudioStreamOpenSLES(const AudioStreamBuilder &builder) + : AudioStreamBuffered(builder) { + // OpenSL ES does not support device IDs. So overwrite value from builder. + mDeviceId = kUnspecified; + // OpenSL ES does not support session IDs. So overwrite value from builder. + mSessionId = SessionId::None; +} + +static constexpr int32_t kFramesPerHighLatencyBurst = 960; // typical, 20 msec at 48000 +static constexpr SLuint32 kAudioChannelCountMax = 30; // TODO Why 30? +static constexpr SLuint32 SL_ANDROID_UNKNOWN_CHANNELMASK = 0; // Matches name used internally. + +SLuint32 AudioStreamOpenSLES::channelCountToChannelMaskDefault(int channelCount) const { + if (channelCount > kAudioChannelCountMax) { + return SL_ANDROID_UNKNOWN_CHANNELMASK; + } + + SLuint32 bitfield = (1 << channelCount) - 1; + + // Check for OS at run-time. + if(getSdkVersion() >= __ANDROID_API_N__) { + return SL_ANDROID_MAKE_INDEXED_CHANNEL_MASK(bitfield); + } + + // Indexed channels masks were added in N. + // For before N, the best we can do is use a positional channel mask. + return bitfield; +} + +static bool s_isLittleEndian() { + static uint32_t value = 1; + return (*reinterpret_cast(&value) == 1); // Does address point to LSB? +} + +SLuint32 AudioStreamOpenSLES::getDefaultByteOrder() { + return s_isLittleEndian() ? SL_BYTEORDER_LITTLEENDIAN : SL_BYTEORDER_BIGENDIAN; +} + +Result AudioStreamOpenSLES::open() { + + LOGI("AudioStreamOpenSLES::open(chans:%d, rate:%d)", + mChannelCount, mSampleRate); + + SLresult result = EngineOpenSLES::getInstance().open(); + if (SL_RESULT_SUCCESS != result) { + return Result::ErrorInternal; + } + + Result oboeResult = AudioStreamBuffered::open(); + if (oboeResult != Result::OK) { + return oboeResult; + } + // Convert to defaults if UNSPECIFIED + if (mSampleRate == kUnspecified) { + mSampleRate = DefaultStreamValues::SampleRate; + } + if (mChannelCount == kUnspecified) { + mChannelCount = DefaultStreamValues::ChannelCount; + } + + mSharingMode = SharingMode::Shared; + + return Result::OK; +} + +Result AudioStreamOpenSLES::configureBufferSizes() { + // Decide frames per burst based on hints from caller. + mFramesPerBurst = mFramesPerCallback; + if (mFramesPerBurst == kUnspecified) { + mFramesPerBurst = DefaultStreamValues::FramesPerBurst; + } + // For high latency streams, use a larger buffer size. + // Performance Mode support was added in N_MR1 (7.1) + if (getSdkVersion() >= __ANDROID_API_N_MR1__ + && mPerformanceMode != PerformanceMode::LowLatency + && mFramesPerBurst < kFramesPerHighLatencyBurst) { + // Find a multiple of framesPerBurst >= kFramesPerHighLatencyBurst. + int32_t numBursts = (kFramesPerHighLatencyBurst + mFramesPerBurst - 1) / mFramesPerBurst; + mFramesPerBurst *= numBursts; + LOGD("AudioStreamOpenSLES:%s() NOT low latency, set mFramesPerBurst = %d", + __func__, mFramesPerBurst); + } + mFramesPerCallback = mFramesPerBurst; + + mBytesPerCallback = mFramesPerCallback * getBytesPerFrame(); + if (mBytesPerCallback <= 0) { + LOGE("AudioStreamOpenSLES::open() bytesPerCallback < 0 = %d, bad format?", + mBytesPerCallback); + return Result::ErrorInvalidFormat; // causing bytesPerFrame == 0 + } + + mCallbackBuffer = std::make_unique(mBytesPerCallback); + + if (!usingFIFO()) { + mBufferCapacityInFrames = mFramesPerBurst * kBufferQueueLength; + mBufferSizeInFrames = mBufferCapacityInFrames; + } + + return Result::OK; +} + +SLuint32 AudioStreamOpenSLES::convertPerformanceMode(PerformanceMode oboeMode) const { + SLuint32 openslMode = SL_ANDROID_PERFORMANCE_NONE; + switch(oboeMode) { + case PerformanceMode::None: + openslMode = SL_ANDROID_PERFORMANCE_NONE; + break; + case PerformanceMode::LowLatency: + openslMode = (getSessionId() == SessionId::None) ? SL_ANDROID_PERFORMANCE_LATENCY : SL_ANDROID_PERFORMANCE_LATENCY_EFFECTS; + break; + case PerformanceMode::PowerSaving: + openslMode = SL_ANDROID_PERFORMANCE_POWER_SAVING; + break; + default: + break; + } + return openslMode; +} + +PerformanceMode AudioStreamOpenSLES::convertPerformanceMode(SLuint32 openslMode) const { + PerformanceMode oboeMode = PerformanceMode::None; + switch(openslMode) { + case SL_ANDROID_PERFORMANCE_NONE: + oboeMode = PerformanceMode::None; + break; + case SL_ANDROID_PERFORMANCE_LATENCY: + case SL_ANDROID_PERFORMANCE_LATENCY_EFFECTS: + oboeMode = PerformanceMode::LowLatency; + break; + case SL_ANDROID_PERFORMANCE_POWER_SAVING: + oboeMode = PerformanceMode::PowerSaving; + break; + default: + break; + } + return oboeMode; +} + +void AudioStreamOpenSLES::logUnsupportedAttributes() { + // Log unsupported attributes + // only report if changed from the default + + // Device ID + if (mDeviceId != kUnspecified) { + LOGW("Device ID [AudioStreamBuilder::setDeviceId()] " + "is not supported on OpenSLES streams."); + } + // Sharing Mode + if (mSharingMode != SharingMode::Shared) { + LOGW("SharingMode [AudioStreamBuilder::setSharingMode()] " + "is not supported on OpenSLES streams."); + } + // Performance Mode + int sdkVersion = getSdkVersion(); + if (mPerformanceMode != PerformanceMode::None && sdkVersion < __ANDROID_API_N_MR1__) { + LOGW("PerformanceMode [AudioStreamBuilder::setPerformanceMode()] " + "is not supported on OpenSLES streams running on pre-Android N-MR1 versions."); + } + // Content Type + if (mContentType != ContentType::Music) { + LOGW("ContentType [AudioStreamBuilder::setContentType()] " + "is not supported on OpenSLES streams."); + } + + // Session Id + if (mSessionId != SessionId::None) { + LOGW("SessionId [AudioStreamBuilder::setSessionId()] " + "is not supported on OpenSLES streams."); + } + + // Input Preset + if (mInputPreset != InputPreset::VoiceRecognition) { + LOGW("InputPreset [AudioStreamBuilder::setInputPreset()] " + "is not supported on OpenSLES streams."); + } +} + +SLresult AudioStreamOpenSLES::configurePerformanceMode(SLAndroidConfigurationItf configItf) { + + if (configItf == nullptr) { + LOGW("%s() called with NULL configuration", __func__); + mPerformanceMode = PerformanceMode::None; + return SL_RESULT_INTERNAL_ERROR; + } + if (getSdkVersion() < __ANDROID_API_N_MR1__) { + LOGW("%s() not supported until N_MR1", __func__); + mPerformanceMode = PerformanceMode::None; + return SL_RESULT_SUCCESS; + } + + SLresult result = SL_RESULT_SUCCESS; + SLuint32 performanceMode = convertPerformanceMode(getPerformanceMode()); + result = (*configItf)->SetConfiguration(configItf, SL_ANDROID_KEY_PERFORMANCE_MODE, + &performanceMode, sizeof(performanceMode)); + if (SL_RESULT_SUCCESS != result) { + LOGW("SetConfiguration(PERFORMANCE_MODE, SL %u) returned %s", + performanceMode, getSLErrStr(result)); + mPerformanceMode = PerformanceMode::None; + } + + return result; +} + +SLresult AudioStreamOpenSLES::updateStreamParameters(SLAndroidConfigurationItf configItf) { + SLresult result = SL_RESULT_SUCCESS; + if(getSdkVersion() >= __ANDROID_API_N_MR1__ && configItf != nullptr) { + SLuint32 performanceMode = 0; + SLuint32 performanceModeSize = sizeof(performanceMode); + result = (*configItf)->GetConfiguration(configItf, SL_ANDROID_KEY_PERFORMANCE_MODE, + &performanceModeSize, &performanceMode); + // A bug in GetConfiguration() before P caused a wrong result code to be returned. + if (getSdkVersion() <= __ANDROID_API_O_MR1__) { + result = SL_RESULT_SUCCESS; // Ignore actual result before P. + } + + if (SL_RESULT_SUCCESS != result) { + LOGW("GetConfiguration(SL_ANDROID_KEY_PERFORMANCE_MODE) returned %d", result); + mPerformanceMode = PerformanceMode::None; // If we can't query it then assume None. + } else { + mPerformanceMode = convertPerformanceMode(performanceMode); // convert SL to Oboe mode + } + } else { + mPerformanceMode = PerformanceMode::None; // If we can't query it then assume None. + } + return result; +} + +Result AudioStreamOpenSLES::close() { + if (mState == StreamState::Closed) { + return Result::ErrorClosed; + } + + AudioStreamBuffered::close(); + + onBeforeDestroy(); + + if (mObjectInterface != nullptr) { + (*mObjectInterface)->Destroy(mObjectInterface); + mObjectInterface = nullptr; + } + + onAfterDestroy(); + + mSimpleBufferQueueInterface = nullptr; + EngineOpenSLES::getInstance().close(); + + setState(StreamState::Closed); + return Result::OK; +} + +SLresult AudioStreamOpenSLES::enqueueCallbackBuffer(SLAndroidSimpleBufferQueueItf bq) { + return (*bq)->Enqueue(bq, mCallbackBuffer.get(), mBytesPerCallback); +} + +int32_t AudioStreamOpenSLES::getBufferDepth(SLAndroidSimpleBufferQueueItf bq) { + SLAndroidSimpleBufferQueueState queueState; + SLresult result = (*bq)->GetState(bq, &queueState); + return (result == SL_RESULT_SUCCESS) ? queueState.count : -1; +} + +void AudioStreamOpenSLES::processBufferCallback(SLAndroidSimpleBufferQueueItf bq) { + bool stopStream = false; + // Ask the callback to fill the output buffer with data. + DataCallbackResult result = fireDataCallback(mCallbackBuffer.get(), mFramesPerCallback); + if (result == DataCallbackResult::Continue) { + // Update Oboe service position based on OpenSL ES position. + updateServiceFrameCounter(); + // Update Oboe client position with frames handled by the callback. + if (getDirection() == Direction::Input) { + mFramesRead += mFramesPerCallback; + } else { + mFramesWritten += mFramesPerCallback; + } + // Pass the data to OpenSLES. + SLresult enqueueResult = enqueueCallbackBuffer(bq); + if (enqueueResult != SL_RESULT_SUCCESS) { + LOGE("%s() returned %d", __func__, enqueueResult); + stopStream = true; + } + } else if (result == DataCallbackResult::Stop) { + LOGD("Oboe callback returned Stop"); + stopStream = true; + } else { + LOGW("Oboe callback returned unexpected value = %d", result); + stopStream = true; + } + if (stopStream) { + requestStop(); + } +} + +// this callback handler is called every time a buffer needs processing +static void bqCallbackGlue(SLAndroidSimpleBufferQueueItf bq, void *context) { + (reinterpret_cast(context))->processBufferCallback(bq); +} + +SLresult AudioStreamOpenSLES::registerBufferQueueCallback() { + // The BufferQueue + SLresult result = (*mObjectInterface)->GetInterface(mObjectInterface, SL_IID_ANDROIDSIMPLEBUFFERQUEUE, + &mSimpleBufferQueueInterface); + if (SL_RESULT_SUCCESS != result) { + LOGE("get buffer queue interface:%p result:%s", + mSimpleBufferQueueInterface, + getSLErrStr(result)); + } else { + // Register the BufferQueue callback + result = (*mSimpleBufferQueueInterface)->RegisterCallback(mSimpleBufferQueueInterface, + bqCallbackGlue, this); + if (SL_RESULT_SUCCESS != result) { + LOGE("RegisterCallback result:%s", getSLErrStr(result)); + } + } + return result; +} + +int32_t AudioStreamOpenSLES::getFramesPerBurst() { + return mFramesPerBurst; +} + +int64_t AudioStreamOpenSLES::getFramesProcessedByServer() const { + int64_t millis64 = mPositionMillis.get(); + int64_t framesProcessed = millis64 * getSampleRate() / kMillisPerSecond; + return framesProcessed; +} + +Result AudioStreamOpenSLES::waitForStateChange(StreamState currentState, + StreamState *nextState, + int64_t timeoutNanoseconds) { + Result oboeResult = Result::ErrorTimeout; + int64_t sleepTimeNanos = 20 * kNanosPerMillisecond; // arbitrary + int64_t timeLeftNanos = timeoutNanoseconds; + + while (true) { + const StreamState state = getState(); // this does not require a lock + if (nextState != nullptr) { + *nextState = state; + } + if (currentState != state) { // state changed? + oboeResult = Result::OK; + break; + } + + // Did we timeout or did user ask for non-blocking? + if (timeoutNanoseconds <= 0) { + break; + } + + if (sleepTimeNanos > timeLeftNanos){ + sleepTimeNanos = timeLeftNanos; + } + AudioClock::sleepForNanos(sleepTimeNanos); + timeLeftNanos -= sleepTimeNanos; + } + + return oboeResult; +} diff --git a/modules/juce_audio_devices/native/oboe/src/opensles/AudioStreamOpenSLES.h b/modules/juce_audio_devices/native/oboe/src/opensles/AudioStreamOpenSLES.h new file mode 100644 index 0000000000..9980c5e412 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/opensles/AudioStreamOpenSLES.h @@ -0,0 +1,132 @@ +/* + * Copyright 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_AUDIO_STREAM_OPENSL_ES_H_ +#define OBOE_AUDIO_STREAM_OPENSL_ES_H_ + +#include + +#include +#include + +#include "oboe/Oboe.h" +#include "common/MonotonicCounter.h" +#include "opensles/AudioStreamBuffered.h" +#include "opensles/EngineOpenSLES.h" + +namespace oboe { + +constexpr int kBitsPerByte = 8; +constexpr int kBufferQueueLength = 2; // double buffered for callbacks + +/** + * INTERNAL USE ONLY + * + * A stream that wraps OpenSL ES. + * + * Do not instantiate this class directly. + * Use an OboeStreamBuilder to create one. + */ + +class AudioStreamOpenSLES : public AudioStreamBuffered { +public: + + AudioStreamOpenSLES(); + explicit AudioStreamOpenSLES(const AudioStreamBuilder &builder); + + virtual ~AudioStreamOpenSLES() = default; + + virtual Result open() override; + virtual Result close() override; + + /** + * Query the current state, eg. OBOE_STREAM_STATE_PAUSING + * + * @return state or a negative error. + */ + StreamState getState() const override { return mState.load(); } + + int32_t getFramesPerBurst() override; + + + AudioApi getAudioApi() const override { + return AudioApi::OpenSLES; + } + + /** + * Process next OpenSL ES buffer. + * Called by by OpenSL ES framework. + * + * This is public, but don't call it directly. + */ + void processBufferCallback(SLAndroidSimpleBufferQueueItf bq); + + Result waitForStateChange(StreamState currentState, + StreamState *nextState, + int64_t timeoutNanoseconds) override; + +protected: + + SLuint32 channelCountToChannelMaskDefault(int channelCount) const; + + virtual Result onBeforeDestroy() { return Result::OK; } + virtual Result onAfterDestroy() { return Result::OK; } + + static SLuint32 getDefaultByteOrder(); + + SLresult registerBufferQueueCallback(); + + int32_t getBufferDepth(SLAndroidSimpleBufferQueueItf bq); + + SLresult enqueueCallbackBuffer(SLAndroidSimpleBufferQueueItf bq); + + SLresult configurePerformanceMode(SLAndroidConfigurationItf configItf); + + SLresult updateStreamParameters(SLAndroidConfigurationItf configItf); + + PerformanceMode convertPerformanceMode(SLuint32 openslMode) const; + SLuint32 convertPerformanceMode(PerformanceMode oboeMode) const; + + Result configureBufferSizes(); + + void logUnsupportedAttributes(); + + /** + * Internal use only. + * Use this instead of directly setting the internal state variable. + */ + void setState(StreamState state) { + mState.store(state); + } + + int64_t getFramesProcessedByServer() const; + + // OpenSLES stuff + SLObjectItf mObjectInterface = nullptr; + SLAndroidSimpleBufferQueueItf mSimpleBufferQueueInterface = nullptr; + + int32_t mBytesPerCallback = oboe::kUnspecified; + MonotonicCounter mPositionMillis; // for tracking OpenSL ES service position + +private: + std::unique_ptr mCallbackBuffer; + std::atomic mState{StreamState::Uninitialized}; + +}; + +} // namespace oboe + +#endif // OBOE_AUDIO_STREAM_OPENSL_ES_H_ diff --git a/modules/juce_audio_devices/native/oboe/src/opensles/EngineOpenSLES.cpp b/modules/juce_audio_devices/native/oboe/src/opensles/EngineOpenSLES.cpp new file mode 100644 index 0000000000..d82219ee09 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/opensles/EngineOpenSLES.cpp @@ -0,0 +1,101 @@ +/* + * Copyright 2017 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include "common/OboeDebug.h" +#include "EngineOpenSLES.h" +#include "OpenSLESUtilities.h" + +using namespace oboe; + +EngineOpenSLES &EngineOpenSLES::getInstance() { + static EngineOpenSLES sInstance; + return sInstance; +} + +SLresult EngineOpenSLES::open() { + std::lock_guard lock(mLock); + + SLresult result = SL_RESULT_SUCCESS; + if (mOpenCount++ == 0) { + + // create engine + result = slCreateEngine(&mEngineObject, 0, NULL, 0, NULL, NULL); + if (SL_RESULT_SUCCESS != result) { + LOGE("EngineOpenSLES - slCreateEngine() result:%s", getSLErrStr(result)); + goto error; + } + + // realize the engine + result = (*mEngineObject)->Realize(mEngineObject, SL_BOOLEAN_FALSE); + if (SL_RESULT_SUCCESS != result) { + LOGE("EngineOpenSLES - Realize() engine result:%s", getSLErrStr(result)); + goto error; + } + + // get the engine interface, which is needed in order to create other objects + result = (*mEngineObject)->GetInterface(mEngineObject, SL_IID_ENGINE, &mEngineInterface); + if (SL_RESULT_SUCCESS != result) { + LOGE("EngineOpenSLES - GetInterface() engine result:%s", getSLErrStr(result)); + goto error; + } + } + + return result; + +error: + close(); + return result; +} + +void EngineOpenSLES::close() { + std::lock_guard lock(mLock); + if (--mOpenCount == 0) { + if (mEngineObject != nullptr) { + (*mEngineObject)->Destroy(mEngineObject); + mEngineObject = nullptr; + mEngineInterface = nullptr; + } + } +} + +SLresult EngineOpenSLES::createOutputMix(SLObjectItf *objectItf) { + return (*mEngineInterface)->CreateOutputMix(mEngineInterface, objectItf, 0, 0, 0); +} + +SLresult EngineOpenSLES::createAudioPlayer(SLObjectItf *objectItf, + SLDataSource *audioSource, + SLDataSink *audioSink) { + + const SLInterfaceID ids[] = {SL_IID_BUFFERQUEUE, SL_IID_ANDROIDCONFIGURATION}; + const SLboolean reqs[] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE}; + + return (*mEngineInterface)->CreateAudioPlayer(mEngineInterface, objectItf, audioSource, + audioSink, + sizeof(ids) / sizeof(ids[0]), ids, reqs); +} + +SLresult EngineOpenSLES::createAudioRecorder(SLObjectItf *objectItf, + SLDataSource *audioSource, + SLDataSink *audioSink) { + + const SLInterfaceID ids[] = {SL_IID_ANDROIDSIMPLEBUFFERQUEUE, SL_IID_ANDROIDCONFIGURATION }; + const SLboolean reqs[] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE}; + + return (*mEngineInterface)->CreateAudioRecorder(mEngineInterface, objectItf, audioSource, + audioSink, + sizeof(ids) / sizeof(ids[0]), ids, reqs); +} + diff --git a/modules/juce_audio_devices/native/oboe/src/opensles/EngineOpenSLES.h b/modules/juce_audio_devices/native/oboe/src/opensles/EngineOpenSLES.h new file mode 100644 index 0000000000..3d238a8c09 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/opensles/EngineOpenSLES.h @@ -0,0 +1,65 @@ +/* + * Copyright 2017 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_ENGINE_OPENSLES_H +#define OBOE_ENGINE_OPENSLES_H + +#include +#include + +#include +#include + +namespace oboe { + +/** + * INTERNAL USE ONLY + */ +class EngineOpenSLES { +public: + static EngineOpenSLES &getInstance(); + + SLresult open(); + + void close(); + + SLresult createOutputMix(SLObjectItf *objectItf); + + SLresult createAudioPlayer(SLObjectItf *objectItf, + SLDataSource *audioSource, + SLDataSink *audioSink); + SLresult createAudioRecorder(SLObjectItf *objectItf, + SLDataSource *audioSource, + SLDataSink *audioSink); + +private: + // Make this a safe Singleton + EngineOpenSLES()= default; + ~EngineOpenSLES()= default; + EngineOpenSLES(const EngineOpenSLES&)= delete; + EngineOpenSLES& operator=(const EngineOpenSLES&)= delete; + + std::mutex mLock; + int32_t mOpenCount = 0; + + SLObjectItf mEngineObject = nullptr; + SLEngineItf mEngineInterface = nullptr; +}; + +} // namespace oboe + + +#endif //OBOE_ENGINE_OPENSLES_H diff --git a/modules/juce_audio_devices/native/oboe/src/opensles/OpenSLESUtilities.cpp b/modules/juce_audio_devices/native/oboe/src/opensles/OpenSLESUtilities.cpp new file mode 100644 index 0000000000..071c0d08f6 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/opensles/OpenSLESUtilities.cpp @@ -0,0 +1,93 @@ +/* + * Copyright 2017 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include "OpenSLESUtilities.h" + +namespace oboe { + +/* + * OSLES Helpers + */ + +const char *getSLErrStr(SLresult code) { + switch (code) { + case 0: + return "SL_RESULT_SUCCESS"; + case 1: + return "SL_RESULT_PRECONDITIONS_VIOLATE"; + case 2: + return "SL_RESULT_PARAMETER_INVALID"; + case 3: + return "SL_RESULT_MEMORY_FAILURE"; + case 4: + return "SL_RESULT_RESOURCE_ERROR"; + case 5: + return "SL_RESULT_RESOURCE_LOST"; + case 6: + return "SL_RESULT_IO_ERROR"; + case 7: + return "SL_RESULT_BUFFER_INSUFFICIENT"; + case 8: + return "SL_RESULT_CONTENT_CORRUPTED"; + case 9: + return "SL_RESULT_CONTENT_UNSUPPORTED"; + case 10: + return "SL_RESULT_CONTENT_NOT_FOUND"; + case 11: + return "SL_RESULT_PERMISSION_DENIED"; + case 12: + return "SL_RESULT_FEATURE_UNSUPPORTED"; + case 13: + return "SL_RESULT_INTERNAL_ERROR"; + case 14: + return "SL_RESULT_UNKNOWN_ERROR"; + case 15: + return "SL_RESULT_OPERATION_ABORTED"; + case 16: + return "SL_RESULT_CONTROL_LOST"; + default: + return "Unknown error"; + } +} + +SLAndroidDataFormat_PCM_EX OpenSLES_createExtendedFormat( + SLDataFormat_PCM format, SLuint32 representation) { + SLAndroidDataFormat_PCM_EX format_pcm_ex; + format_pcm_ex.formatType = SL_ANDROID_DATAFORMAT_PCM_EX; + format_pcm_ex.numChannels = format.numChannels; + format_pcm_ex.sampleRate = format.samplesPerSec; + format_pcm_ex.bitsPerSample = format.bitsPerSample; + format_pcm_ex.containerSize = format.containerSize; + format_pcm_ex.channelMask = format.channelMask; + format_pcm_ex.endianness = format.endianness; + format_pcm_ex.representation = representation; + return format_pcm_ex; +} + +SLuint32 OpenSLES_ConvertFormatToRepresentation(AudioFormat format) { + switch(format) { + case AudioFormat::I16: + return SL_ANDROID_PCM_REPRESENTATION_SIGNED_INT; + case AudioFormat::Float: + return SL_ANDROID_PCM_REPRESENTATION_FLOAT; + case AudioFormat::Invalid: + case AudioFormat::Unspecified: + default: + return 0; + } +} + +} // namespace oboe diff --git a/modules/juce_audio_devices/native/oboe/src/opensles/OpenSLESUtilities.h b/modules/juce_audio_devices/native/oboe/src/opensles/OpenSLESUtilities.h new file mode 100644 index 0000000000..50c0e2da9e --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/opensles/OpenSLESUtilities.h @@ -0,0 +1,44 @@ +/* + * Copyright 2017 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_OPENSLES_OPENSLESUTILITIES_H +#define OBOE_OPENSLES_OPENSLESUTILITIES_H + +#include +#include "oboe/Oboe.h" + +namespace oboe { + +const char *getSLErrStr(SLresult code); + +/** + * Creates an extended PCM format from the supplied format and data representation. This method + * should only be called on Android devices with API level 21+. API 21 introduced the + * SLAndroidDataFormat_PCM_EX object which allows audio samples to be represented using + * single precision floating-point. + * + * @param format + * @param representation + * @return the extended PCM format + */ +SLAndroidDataFormat_PCM_EX OpenSLES_createExtendedFormat(SLDataFormat_PCM format, + SLuint32 representation); + +SLuint32 OpenSLES_ConvertFormatToRepresentation(AudioFormat format); + +} // namespace oboe + +#endif //OBOE_OPENSLES_OPENSLESUTILITIES_H diff --git a/modules/juce_audio_devices/native/oboe/src/opensles/OutputMixerOpenSLES.cpp b/modules/juce_audio_devices/native/oboe/src/opensles/OutputMixerOpenSLES.cpp new file mode 100644 index 0000000000..e06f306bae --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/opensles/OutputMixerOpenSLES.cpp @@ -0,0 +1,74 @@ +/* + * Copyright 2017 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + + +#include "common/OboeDebug.h" +#include "EngineOpenSLES.h" +#include "OpenSLESUtilities.h" +#include "OutputMixerOpenSLES.h" + +using namespace oboe; + +OutputMixerOpenSL &OutputMixerOpenSL::getInstance() { + static OutputMixerOpenSL sInstance; + return sInstance; +} + +SLresult OutputMixerOpenSL::open() { + std::lock_guard lock(mLock); + + SLresult result = SL_RESULT_SUCCESS; + if (mOpenCount++ == 0) { + // get the output mixer + result = EngineOpenSLES::getInstance().createOutputMix(&mOutputMixObject); + if (SL_RESULT_SUCCESS != result) { + LOGE("OutputMixerOpenSL() - createOutputMix() result:%s", getSLErrStr(result)); + goto error; + } + + // realize the output mix + result = (*mOutputMixObject)->Realize(mOutputMixObject, SL_BOOLEAN_FALSE); + if (SL_RESULT_SUCCESS != result) { + LOGE("OutputMixerOpenSL() - Realize() mOutputMixObject result:%s", getSLErrStr(result)); + goto error; + } + } + + return result; + +error: + close(); + return result; +} + +void OutputMixerOpenSL::close() { + std::lock_guard lock(mLock); + + if (--mOpenCount == 0) { + // destroy output mix object, and invalidate all associated interfaces + if (mOutputMixObject != nullptr) { + (*mOutputMixObject)->Destroy(mOutputMixObject); + mOutputMixObject = nullptr; + } + } +} + +SLresult OutputMixerOpenSL::createAudioPlayer(SLObjectItf *objectItf, + SLDataSource *audioSource) { + SLDataLocator_OutputMix loc_outmix = {SL_DATALOCATOR_OUTPUTMIX, mOutputMixObject}; + SLDataSink audioSink = {&loc_outmix, NULL}; + return EngineOpenSLES::getInstance().createAudioPlayer(objectItf, audioSource, &audioSink); +} diff --git a/modules/juce_audio_devices/native/oboe/src/opensles/OutputMixerOpenSLES.h b/modules/juce_audio_devices/native/oboe/src/opensles/OutputMixerOpenSLES.h new file mode 100644 index 0000000000..813fd01881 --- /dev/null +++ b/modules/juce_audio_devices/native/oboe/src/opensles/OutputMixerOpenSLES.h @@ -0,0 +1,58 @@ +/* + * Copyright 2017 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef OBOE_OUTPUT_MIXER_OPENSLES_H +#define OBOE_OUTPUT_MIXER_OPENSLES_H + +#include +#include + +#include +#include + +namespace oboe { + +/** + * INTERNAL USE ONLY + */ + +class OutputMixerOpenSL { +public: + static OutputMixerOpenSL &getInstance(); + + SLresult open(); + + void close(); + + SLresult createAudioPlayer(SLObjectItf *objectItf, + SLDataSource *audioSource); + +private: + // Make this a safe Singleton + OutputMixerOpenSL()= default; + ~OutputMixerOpenSL()= default; + OutputMixerOpenSL(const OutputMixerOpenSL&)= delete; + OutputMixerOpenSL& operator=(const OutputMixerOpenSL&)= delete; + + std::mutex mLock; + int32_t mOpenCount = 0; + + SLObjectItf mOutputMixObject = nullptr; +}; + +} // namespace oboe + +#endif //OBOE_OUTPUT_MIXER_OPENSLES_H