349 lines
12 KiB
C++
349 lines
12 KiB
C++
/******************************************************************************\
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* Copyright (c) 2004-2016
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*
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* Author(s):
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* Volker Fischer
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*
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******************************************************************************
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*
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* This program is free software; you can redistribute it and/or modify it under
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* the terms of the GNU General Public License as published by the Free Software
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* Foundation; either version 2 of the License, or (at your option) any later
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* version.
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*
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* This program is distributed in the hope that it will be useful, but WITHOUT
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* ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
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* FOR A PARTICULAR PURPOSE. See the GNU General Public License for more
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* details.
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*
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* You should have received a copy of the GNU General Public License along with
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* this program; if not, write to the Free Software Foundation, Inc.,
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* 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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*
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\******************************************************************************/
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#include "sound.h"
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/* Implementation *************************************************************/
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CSound::CSound ( void (*fpNewProcessCallback) ( CVector<short>& psData, void* arg ), void* arg ) :
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CSoundBase ( "OpenSL", true, fpNewProcessCallback, arg )
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{
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}
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void CSound::InitializeOpenSL()
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{
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// set up stream formats for input and output
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SLDataFormat_PCM inStreamFormat;
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inStreamFormat.formatType = SL_DATAFORMAT_PCM;
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inStreamFormat.numChannels = 1;
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inStreamFormat.samplesPerSec = SL_SAMPLINGRATE_16;
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inStreamFormat.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16;
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inStreamFormat.containerSize = 16;
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inStreamFormat.channelMask = SL_SPEAKER_FRONT_CENTER;
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inStreamFormat.endianness = SL_BYTEORDER_LITTLEENDIAN;
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SLDataFormat_PCM outStreamFormat;
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outStreamFormat.formatType = SL_DATAFORMAT_PCM;
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outStreamFormat.numChannels = 2;
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outStreamFormat.samplesPerSec = SYSTEM_SAMPLE_RATE_HZ * 1000; // unit is mHz
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outStreamFormat.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16;
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outStreamFormat.containerSize = 16;
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outStreamFormat.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
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outStreamFormat.endianness = SL_BYTEORDER_LITTLEENDIAN;
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// create the OpenSL root engine object
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slCreateEngine ( &engineObject,
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0,
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nullptr,
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0,
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nullptr,
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nullptr );
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// realize the engine
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(*engineObject)->Realize ( engineObject,
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SL_BOOLEAN_FALSE );
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// get the engine interface (required to create other objects)
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(*engineObject)->GetInterface ( engineObject,
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SL_IID_ENGINE,
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&engine );
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// create the main output mix
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(*engine)->CreateOutputMix ( engine,
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&outputMixObject,
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0,
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nullptr,
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nullptr );
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// realize the output mix
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(*outputMixObject)->Realize ( outputMixObject,
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SL_BOOLEAN_FALSE );
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// configure the audio (data) source for input
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SLDataLocator_IODevice micLocator;
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micLocator.locatorType = SL_DATALOCATOR_IODEVICE;
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micLocator.deviceType = SL_IODEVICE_AUDIOINPUT;
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micLocator.deviceID = SL_DEFAULTDEVICEID_AUDIOINPUT;
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micLocator.device = nullptr;
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SLDataSource inDataSource;
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inDataSource.pLocator = &micLocator;
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inDataSource.pFormat = nullptr;
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// configure the input buffer queue
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SLDataLocator_AndroidSimpleBufferQueue inBufferQueue;
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inBufferQueue.locatorType = SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE;
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inBufferQueue.numBuffers = 2; // max number of buffers in queue
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// configure the audio (data) sink for input
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SLDataSink inDataSink;
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inDataSink.pLocator = &inBufferQueue;
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inDataSink.pFormat = &inStreamFormat;
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// create the audio recorder
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const SLInterfaceID recorderIds[] = { SL_IID_ANDROIDSIMPLEBUFFERQUEUE };
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const SLboolean recorderReq[] = { SL_BOOLEAN_TRUE };
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(*engine)->CreateAudioRecorder ( engine,
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&recorderObject,
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&inDataSource,
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&inDataSink,
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1,
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recorderIds,
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recorderReq );
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// realize the audio recorder
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(*recorderObject)->Realize ( recorderObject,
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SL_BOOLEAN_FALSE );
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// get the audio recorder interface
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(*recorderObject)->GetInterface ( recorderObject,
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SL_IID_RECORD,
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&recorder );
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// get the audio recorder simple buffer queue interface
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(*recorderObject)->GetInterface ( recorderObject,
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SL_IID_ANDROIDSIMPLEBUFFERQUEUE,
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&recorderSimpleBufQueue );
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// register the audio input callback
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(*recorderSimpleBufQueue)->RegisterCallback ( recorderSimpleBufQueue,
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processInput,
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this );
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// configure the output buffer queue
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SLDataLocator_AndroidSimpleBufferQueue outBufferQueue;
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outBufferQueue.locatorType = SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE;
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outBufferQueue.numBuffers = 2; // max number of buffers in queue
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// configure the audio (data) source for output
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SLDataSource outDataSource;
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outDataSource.pLocator = &outBufferQueue;
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outDataSource.pFormat = &outStreamFormat;
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// configure the output mix
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SLDataLocator_OutputMix outputMix;
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outputMix.locatorType = SL_DATALOCATOR_OUTPUTMIX;
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outputMix.outputMix = outputMixObject;
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// configure the audio (data) sink for output
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SLDataSink outDataSink;
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outDataSink.pLocator = &outputMix;
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outDataSink.pFormat = nullptr;
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// create the audio player
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const SLInterfaceID playerIds[] = { SL_IID_ANDROIDSIMPLEBUFFERQUEUE };
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const SLboolean playerReq[] = { SL_BOOLEAN_TRUE };
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(*engine)->CreateAudioPlayer ( engine,
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&playerObject,
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&outDataSource,
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&outDataSink,
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1,
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playerIds,
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playerReq );
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// realize the audio player
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(*playerObject)->Realize ( playerObject,
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SL_BOOLEAN_FALSE );
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// get the audio player interface
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(*playerObject)->GetInterface ( playerObject,
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SL_IID_PLAY,
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&player );
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// get the audio player simple buffer queue interface
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(*playerObject)->GetInterface ( playerObject,
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SL_IID_ANDROIDSIMPLEBUFFERQUEUE,
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&playerSimpleBufQueue );
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// register the audio output callback
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(*playerSimpleBufQueue)->RegisterCallback ( playerSimpleBufQueue,
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processOutput,
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this );
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}
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void CSound::CloseOpenSL()
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{
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// clean up
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(*recorderObject)->Destroy ( recorderObject );
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(*playerObject)->Destroy ( playerObject );
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(*outputMixObject)->Destroy ( outputMixObject );
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(*engineObject)->Destroy ( engineObject );
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}
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void CSound::Start()
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{
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InitializeOpenSL();
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// TEST We have to supply the interface with initial buffers, otherwise
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// the rendering will not start.
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// Note that the number of buffers enqueued here must match the maximum
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// numbers of buffers configured in the constructor of this class.
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vecsTmpAudioSndCrdStereo.Reset ( 0 );
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// enqueue initial buffers for record
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(*recorderSimpleBufQueue)->Enqueue ( recorderSimpleBufQueue,
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&vecsTmpAudioSndCrdStereo[0],
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iOpenSLBufferSizeStereo * 2 /* 2 bytes */ );
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(*recorderSimpleBufQueue)->Enqueue ( recorderSimpleBufQueue,
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&vecsTmpAudioSndCrdStereo[0],
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iOpenSLBufferSizeStereo * 2 /* 2 bytes */ );
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// enqueue initial buffers for playback
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(*playerSimpleBufQueue)->Enqueue ( playerSimpleBufQueue,
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&vecsTmpAudioSndCrdStereo[0],
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iOpenSLBufferSizeStereo * 2 /* 2 bytes */ );
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(*playerSimpleBufQueue)->Enqueue ( playerSimpleBufQueue,
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&vecsTmpAudioSndCrdStereo[0],
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iOpenSLBufferSizeStereo * 2 /* 2 bytes */ );
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// start the rendering
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(*recorder)->SetRecordState ( recorder, SL_RECORDSTATE_RECORDING );
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(*player)->SetPlayState ( player, SL_PLAYSTATE_PLAYING );
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// call base class
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CSoundBase::Start();
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}
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void CSound::Stop()
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{
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// stop the audio stream
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(*recorder)->SetRecordState ( recorder, SL_RECORDSTATE_STOPPED );
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(*player)->SetPlayState ( player, SL_PLAYSTATE_STOPPED );
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// clear the buffers
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(*recorderSimpleBufQueue)->Clear ( recorderSimpleBufQueue );
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(*playerSimpleBufQueue)->Clear ( playerSimpleBufQueue );
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// call base class
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CSoundBase::Stop();
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CloseOpenSL();
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}
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int CSound::Init ( const int iNewPrefMonoBufferSize )
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{
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// TODO make use of the following:
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// String sampleRate = am.getProperty(AudioManager.PROPERTY_OUTPUT_SAMPLE_RATE));
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// String framesPerBuffer = am.getProperty(AudioManager.PROPERTY_OUTPUT_FRAMES_PER_BUFFER));
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/*
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// get the Audio IO DEVICE CAPABILITIES interface
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SLAudioIODeviceCapabilitiesItf audioCapabilities;
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(*engineObject)->GetInterface ( engineObject,
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SL_IID_AUDIOIODEVICECAPABILITIES,
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&audioCapabilities );
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(*audioCapabilities)->QueryAudioInputCapabilities ( audioCapabilities,
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inputDeviceIDs[i],
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&audioInputDescriptor );
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*/
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// store buffer size
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iOpenSLBufferSizeMono = iNewPrefMonoBufferSize;
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// init base class
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CSoundBase::Init ( iOpenSLBufferSizeMono );
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// set internal buffer size value and calculate stereo buffer size
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iOpenSLBufferSizeStereo = 2 * iOpenSLBufferSizeMono;
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// create memory for intermediate audio buffer
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vecsTmpAudioSndCrdStereo.Init ( iOpenSLBufferSizeStereo );
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// TEST
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#if ( SYSTEM_SAMPLE_RATE_HZ != 48000 )
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# error "Only a system sample rate of 48 kHz is supported by this module"
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#endif
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// audio interface number of channels is 1 and the sample rate
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// is 16 kHz -> just copy samples and perform no filtering as a
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// first simple solution
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// 48 kHz / 16 kHz = factor 3 (note that the buffer size mono might
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// be divisible by three, therefore we will get a lot of drop outs)
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iModifiedInBufSize = iOpenSLBufferSizeMono / 3;
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vecsTmpAudioInSndCrd.Init ( iModifiedInBufSize );
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return iOpenSLBufferSizeMono;
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}
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void CSound::processInput ( SLAndroidSimpleBufferQueueItf bufferQueue,
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void* instance )
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{
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CSound* pSound = static_cast<CSound*> ( instance );
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// only process if we are running
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if ( !pSound->bRun )
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{
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return;
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}
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QMutexLocker locker ( &pSound->Mutex );
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// enqueue the buffer for record
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(*bufferQueue)->Enqueue ( bufferQueue,
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&pSound->vecsTmpAudioInSndCrd[0],
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pSound->iModifiedInBufSize * 2 /* 2 bytes */ );
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// upsampling (without filtering) and channel management
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pSound->vecsTmpAudioSndCrdStereo.Reset ( 0 );
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for ( int i = 0; i < pSound->iModifiedInBufSize; i++ )
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{
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pSound->vecsTmpAudioSndCrdStereo[6 * i] =
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pSound->vecsTmpAudioSndCrdStereo[6 * i + 1] =
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pSound->vecsTmpAudioInSndCrd[i];
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}
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}
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void CSound::processOutput ( SLAndroidSimpleBufferQueueItf bufferQueue,
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void* instance )
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{
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CSound* pSound = static_cast<CSound*> ( instance );
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// only process if we are running
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if ( !pSound->bRun )
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{
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return;
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}
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QMutexLocker locker ( &pSound->Mutex );
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// call processing callback function
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pSound->ProcessCallback ( pSound->vecsTmpAudioSndCrdStereo );
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// enqueue the buffer for playback
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(*bufferQueue)->Enqueue ( bufferQueue,
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&pSound->vecsTmpAudioSndCrdStereo[0],
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pSound->iOpenSLBufferSizeStereo * 2 /* 2 bytes */ );
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}
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