Java example source code file (PLATFORM_API_MacOSX_PCM.cpp)

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Java - Java tags/keywords

daudio_open, inputcallback, int64, null, os_error0, osstatus, osx_directaudiodevice, osx_directaudiodevice\*, pos2offset, trace0, trace1, trace2, true, uint32

The PLATFORM_API_MacOSX_PCM.cpp Java example source code

/*
 * Copyright (c) 2002, 2012, Oracle and/or its affiliates. All rights reserved.
 * DO NOT ALTER OR REMOVE COPYRIGHT NOTICES OR THIS FILE HEADER.
 *
 * This code is free software; you can redistribute it and/or modify it
 * under the terms of the GNU General Public License version 2 only, as
 * published by the Free Software Foundation.  Oracle designates this
 * particular file as subject to the "Classpath" exception as provided
 * by Oracle in the LICENSE file that accompanied this code.
 *
 * This code is distributed in the hope that it will be useful, but WITHOUT
 * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
 * FITNESS FOR A PARTICULAR PURPOSE.  See the GNU General Public License
 * version 2 for more details (a copy is included in the LICENSE file that
 * accompanied this code).
 *
 * You should have received a copy of the GNU General Public License version
 * 2 along with this work; if not, write to the Free Software Foundation,
 * Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA.
 *
 * Please contact Oracle, 500 Oracle Parkway, Redwood Shores, CA 94065 USA
 * or visit www.oracle.com if you need additional information or have any
 * questions.
 */

//#define USE_ERROR
//#define USE_TRACE
//#define USE_VERBOSE_TRACE

#include <AudioUnit/AudioUnit.h>
#include <CoreServices/CoreServices.h>
#include <AudioToolbox/AudioConverter.h>
#include <pthread.h>
#include <math.h>
/*
#if !defined(__COREAUDIO_USE_FLAT_INCLUDES__)
#include <CoreAudio/CoreAudioTypes.h>
#else
#include <CoreAudioTypes.h>
#endif
*/

#include "PLATFORM_API_MacOSX_Utils.h"

extern "C" {
#include "Utilities.h"
#include "DirectAudio.h"
}

#if USE_DAUDIO == TRUE


#ifdef USE_TRACE
static void PrintStreamDesc(const AudioStreamBasicDescription *inDesc) {
    TRACE4("ID='%c%c%c%c'", (char)(inDesc->mFormatID >> 24), (char)(inDesc->mFormatID >> 16), (char)(inDesc->mFormatID >> 8), (char)(inDesc->mFormatID));
    TRACE2(", %f Hz, flags=0x%lX", (float)inDesc->mSampleRate, (long unsigned)inDesc->mFormatFlags);
    TRACE2(", %ld channels, %ld bits", (long)inDesc->mChannelsPerFrame, (long)inDesc->mBitsPerChannel);
    TRACE1(", %ld bytes per frame\n", (long)inDesc->mBytesPerFrame);
}
#else
static inline void PrintStreamDesc(const AudioStreamBasicDescription *inDesc) { }
#endif


#define MAX(x, y)   ((x) >= (y) ? (x) : (y))
#define MIN(x, y)   ((x) <= (y) ? (x) : (y))


// =======================================
// MixerProvider functions implementation

static DeviceList deviceCache;

INT32 DAUDIO_GetDirectAudioDeviceCount() {
    deviceCache.Refresh();
    int count = deviceCache.GetCount();
    if (count > 0) {
        // add "default" device
        count++;
        TRACE1("DAUDIO_GetDirectAudioDeviceCount: returns %d devices\n", count);
    } else {
        TRACE0("DAUDIO_GetDirectAudioDeviceCount: no devices found\n");
    }
    return count;
}

INT32 DAUDIO_GetDirectAudioDeviceDescription(INT32 mixerIndex, DirectAudioDeviceDescription *desc) {
    bool result = true;
    desc->deviceID = 0;
    if (mixerIndex == 0) {
        // default device
        strncpy(desc->name, "Default Audio Device", DAUDIO_STRING_LENGTH);
        strncpy(desc->description, "Default Audio Device", DAUDIO_STRING_LENGTH);
        desc->maxSimulLines = -1;
    } else {
        AudioDeviceID deviceID;
        result = deviceCache.GetDeviceInfo(mixerIndex-1, &deviceID, DAUDIO_STRING_LENGTH,
            desc->name, desc->vendor, desc->description, desc->version);
        if (result) {
            desc->deviceID = (INT32)deviceID;
            desc->maxSimulLines = -1;
        }
    }
    return result ? TRUE : FALSE;
}


void DAUDIO_GetFormats(INT32 mixerIndex, INT32 deviceID, int isSource, void* creator) {
    TRACE3(">>DAUDIO_GetFormats mixerIndex=%d deviceID=0x%x isSource=%d\n", (int)mixerIndex, (int)deviceID, isSource);

    AudioDeviceID audioDeviceID = deviceID == 0 ? GetDefaultDevice(isSource) : (AudioDeviceID)deviceID;

    if (audioDeviceID == 0) {
        return;
    }

    int totalChannels = GetChannelCount(audioDeviceID, isSource);

    if (totalChannels == 0) {
        TRACE0("<Resampler::Init\n");
        TRACE0("  inFormat: ");
        PrintStreamDesc(inFormat);
        TRACE0("  outFormat: ");
        PrintStreamDesc(outFormat);
        TRACE1("  inputBufferSize: %d bytes\n", inputBufferSizeInBytes);
        OSStatus err;

        if ((outFormat->mFormatFlags & kAudioFormatFlagIsNonInterleaved) != 0 && outFormat->mChannelsPerFrame != 1) {
            ERROR0("Resampler::Init ERROR: outFormat is non-interleaved\n");
            return false;
        }
        if ((inFormat->mFormatFlags & kAudioFormatFlagIsNonInterleaved) != 0 && inFormat->mChannelsPerFrame != 1) {
            ERROR0("Resampler::Init ERROR: inFormat is non-interleaved\n");
            return false;
        }

        memcpy(&asbdIn, inFormat, sizeof(AudioStreamBasicDescription));
        memcpy(&asbdOut, outFormat, sizeof(AudioStreamBasicDescription));

        err = AudioConverterNew(inFormat, outFormat, &converter);

        if (err || converter == NULL) {
            OS_ERROR1(err, "Resampler::Init (AudioConverterNew), converter=%p", converter);
            return false;
        }

        // allocate buffer for output data
        int maximumInFrames = inputBufferSizeInBytes / inFormat->mBytesPerFrame;
        // take into account trailingFrames
        AudioConverterPrimeInfo primeInfo = {0, 0};
        UInt32 sizePrime = sizeof(primeInfo);
        err = AudioConverterGetProperty(converter, kAudioConverterPrimeInfo, &sizePrime, &primeInfo);
        if (err) {
            OS_ERROR0(err, "Resampler::Init (get kAudioConverterPrimeInfo)");
            // ignore the error
        } else {
            // the default primeMethod is kConverterPrimeMethod_Normal, so we need only trailingFrames
            maximumInFrames += primeInfo.trailingFrames;
        }
        float outBufferSizeInFrames = (outFormat->mSampleRate / inFormat->mSampleRate) * ((float)maximumInFrames);
        // to avoid complex calculation just set outBufferSize as double of the calculated value
        outBufferSize = (int)outBufferSizeInFrames * outFormat->mBytesPerFrame * 2;
        // safety check - consider 256 frame as the minimum input buffer
        int minOutSize = 256 * outFormat->mBytesPerFrame;
        if (outBufferSize < minOutSize) {
            outBufferSize = minOutSize;
        }

        outBuffer = malloc(outBufferSize);

        if (outBuffer == NULL) {
            ERROR1("Resampler::Init ERROR: malloc failed (%d bytes)\n", outBufferSize);
            AudioConverterDispose(converter);
            converter = NULL;
            return false;
        }

        TRACE1("  allocated: %d bytes for output buffer\n", outBufferSize);

        TRACE0("<Resampler::Process: %d bytes, converter = %p\n", len, converter);
        if (converter == NULL) {    // sanity check
            bytesWritten = ringBuffer->Write(srcBuffer, len, false);
        } else {
            InputProcData data;
            data.pThis = this;
            data.data = (Byte *)srcBuffer;
            data.dataSize = len;

            OSStatus err;
            do {
                AudioBufferList abl;    // by default it contains 1 AudioBuffer
                abl.mNumberBuffers = 1;
                abl.mBuffers[0].mNumberChannels = asbdOut.mChannelsPerFrame;
                abl.mBuffers[0].mDataByteSize   = outBufferSize;
                abl.mBuffers[0].mData           = outBuffer;

                UInt32 packets = (UInt32)outBufferSize / asbdOut.mBytesPerPacket;

                TRACE2(">>AudioConverterFillComplexBuffer: request %d packets, provide %d bytes buffer\n",
                    (int)packets, (int)abl.mBuffers[0].mDataByteSize);

                err = AudioConverterFillComplexBuffer(converter, ConverterInputProc, &data, &packets, &abl, NULL);

                TRACE2("<Resampler::Discontinue\n");
        if (converter != NULL) {
            AudioConverterReset(converter);
        }
        TRACE0("<ConverterInputProc: requested %d packets, data contains %d bytes (%d packets)\n",
            (int)*ioNumberDataPackets, (int)data->dataSize, (int)(data->dataSize / data->pThis->asbdIn.mBytesPerPacket));
        if (data->dataSize == 0) {
            // already called & provided all input data
            // interrupt conversion by returning error
            *ioNumberDataPackets = 0;
            TRACE0("  <pThis->asbdIn.mChannelsPerFrame;
        ioData->mBuffers[0].mDataByteSize   = data->dataSize;
        ioData->mBuffers[0].mData           = data->data;

        *ioNumberDataPackets = data->dataSize / data->pThis->asbdIn.mBytesPerPacket;

        // all data has been provided to the converter
        data->dataSize = 0;

        TRACE1("  <OutputCallback: busNum=%d, requested %d frames (%d bytes)\n",
        (int)inBusNumber, (int)inNumberFrames, (int)(inNumberFrames * device->asbd.mBytesPerFrame));
    TRACE3("  abl: %d buffers, buffer[0].channels=%d, buffer.size=%d\n",
        nchannels, (int)audioBuffer->mNumberChannels, (int)audioBuffer->mDataByteSize);

    int bytesToRead = inNumberFrames * device->asbd.mBytesPerFrame;
    if (bytesToRead > (int)audioBuffer->mDataByteSize) {
        TRACE0("--OutputCallback: !!! audioBuffer IS TOO SMALL!!!\n");
        bytesToRead = audioBuffer->mDataByteSize / device->asbd.mBytesPerFrame * device->asbd.mBytesPerFrame;
    }
    int bytesRead = device->ringBuffer.Read(audioBuffer->mData, bytesToRead);
    if (bytesRead < bytesToRead) {
        // no enough data (underrun)
        TRACE2("--OutputCallback: !!! UNDERRUN (read %d bytes of %d)!!!\n", bytesRead, bytesToRead);
        // silence the rest
        memset((Byte*)audioBuffer->mData + bytesRead, 0, bytesToRead-bytesRead);
        bytesRead = bytesToRead;
    }

    audioBuffer->mDataByteSize = (UInt32)bytesRead;
    // SAFETY: set mDataByteSize for all other AudioBuffer in the AudioBufferList to zero
    while (--nchannels > 0) {
        audioBuffer++;
        audioBuffer->mDataByteSize = 0;
    }
    TRACE1("<InputCallback: busNum=%d, timeStamp=%lld, %d frames (%d bytes)\n",
        (int)inBusNumber, (long long)inTimeStamp->mSampleTime, (int)inNumberFrames, (int)(inNumberFrames * device->asbd.mBytesPerFrame));

    AudioBufferList abl;    // by default it contains 1 AudioBuffer
    abl.mNumberBuffers = 1;
    abl.mBuffers[0].mNumberChannels = device->asbd.mChannelsPerFrame;
    abl.mBuffers[0].mDataByteSize   = device->inputBufferSizeInBytes;   // assume this is == (inNumberFrames * device->asbd.mBytesPerFrame)
    abl.mBuffers[0].mData           = NULL;     // request for the audioUnit's buffer

    OSStatus err = AudioUnitRender(device->audioUnit, ioActionFlags, inTimeStamp, inBusNumber, inNumberFrames, &abl);
    if (err) {
        OS_ERROR0(err, "<Discontinue();
            } else {
                TRACE2("  InputCallback (RESAMPLED), continuous: lastWrittenSampleTime = %f, sampleTime=%f\n",
                    (float)device->lastWrittenSampleTime, (float)sampleTime);
            }
            device->lastWrittenSampleTime = sampleTime + inNumberFrames;

            int bytesWritten = device->resampler->Process(abl.mBuffers[0].mData, (int)abl.mBuffers[0].mDataByteSize, &device->ringBuffer);
            TRACE2("<DAUDIO_Open: mixerIndex=%d deviceID=0x%x isSource=%d\n", (int)mixerIndex, (unsigned int)deviceID, isSource);
    TRACE3("  sampleRate=%d sampleSizeInBits=%d channels=%d\n", (int)sampleRate, sampleSizeInBits, channels);
#ifdef USE_TRACE
    {
        AudioDeviceID audioDeviceID = deviceID;
        if (audioDeviceID == 0) {
            // default device
            audioDeviceID = GetDefaultDevice(isSource);
        }
        char name[256];
        OSStatus err = GetAudioObjectProperty(audioDeviceID, kAudioUnitScope_Global, kAudioDevicePropertyDeviceName, 256, &name, 0);
        if (err != noErr) {
            OS_ERROR1(err, "  audioDeviceID=0x%x, name is N/A:", (int)audioDeviceID);
        } else {
            TRACE2("  audioDeviceID=0x%x, name=%s\n", (int)audioDeviceID, name);
        }
    }
#endif

    if (encoding != DAUDIO_PCM) {
        ERROR1("<asbd, sizeof(device->asbd));
    if (err) {
        OS_ERROR0(err, "<asbd.mBytesPerFrame;  // convert frames to bytes
        extraBufferBytes = (int)device->inputBufferSizeInBytes;
    }

    if (device->resampler != NULL) {
        // resampler output format is a user requested format (== ringBuffer format)
        AudioStreamBasicDescription asbdOut; // ringBuffer format
        FillASBDForNonInterleavedPCM(asbdOut, sampleRate, channels, sampleSizeInBits, false, isSigned, isBigEndian);

        // set resampler input buffer size to the HAL buffer size
        if (!device->resampler->Init(&device->asbd, &asbdOut, (int)device->inputBufferSizeInBytes)) {
            ERROR0("<GetOutBufferSize();
    }

    if (!device->ringBuffer.Allocate(bufferSizeInBytes, extraBufferBytes)) {
        ERROR0("<DAUDIO_Write: %d bytes to write\n", byteSize);

    int result = device->ringBuffer.Write(data, byteSize, true);

    TRACE1("<DAUDIO_Read: %d bytes to read\n", byteSize);

    int result = device->ringBuffer.Read(data, byteSize);

    TRACE1("< 0 ? TRUE : FALSE;

    TRACE1("DAUDIO_StillDraining returns %d\n", draining);
    return draining;
}

int DAUDIO_Flush(void* id, int isSource) {
    OSX_DirectAudioDevice *device = (OSX_DirectAudioDevice*)id;
    TRACE0("DAUDIO_Flush\n");

    device->ringBuffer.Flush();

    return TRUE;
}

int DAUDIO_GetAvailable(void* id, int isSource) {
    OSX_DirectAudioDevice *device = (OSX_DirectAudioDevice*)id;

    int bytesInBuffer = device->ringBuffer.GetValidByteCount();
    if (isSource) {
        return device->ringBuffer.GetBufferSize() - bytesInBuffer;
    } else {
        return bytesInBuffer;
    }
}

INT64 DAUDIO_GetBytePosition(void* id, int isSource, INT64 javaBytePos) {
    OSX_DirectAudioDevice *device = (OSX_DirectAudioDevice*)id;
    INT64 position;

    if (isSource) {
        position = javaBytePos - device->ringBuffer.GetValidByteCount();
    } else {
        position = javaBytePos + device->ringBuffer.GetValidByteCount();
    }

    TRACE2("DAUDIO_GetBytePosition returns %lld (javaBytePos = %lld)\n", (long long)position, (long long)javaBytePos);
    return position;
}

void DAUDIO_SetBytePosition(void* id, int isSource, INT64 javaBytePos) {
    // no need javaBytePos (it's available in DAUDIO_GetBytePosition)
}

int DAUDIO_RequiresServicing(void* id, int isSource) {
    return FALSE;
}

void DAUDIO_Service(void* id, int isSource) {
    // unreachable
}

#endif  // USE_DAUDIO == TRUE