| /* |
| * Copyright (c) 2002, 2015, 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 <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("<<DAUDIO_GetFormats, no streams!\n"); |
| return; |
| } |
| |
| if (isSource && totalChannels < 2) { |
| // report 2 channels even if only mono is supported |
| totalChannels = 2; |
| } |
| |
| int channels[] = {1, 2, totalChannels}; |
| int channelsCount = MIN(totalChannels, 3); |
| |
| float hardwareSampleRate = GetSampleRate(audioDeviceID, isSource); |
| TRACE2(" DAUDIO_GetFormats: got %d channels, sampleRate == %f\n", totalChannels, hardwareSampleRate); |
| |
| // any sample rates are supported |
| float sampleRate = -1; |
| |
| static int sampleBits[] = {8, 16, 24}; |
| static int sampleBitsCount = sizeof(sampleBits)/sizeof(sampleBits[0]); |
| |
| // the last audio format is the default one (used by DataLine.open() if format is not specified) |
| // consider as default 16bit PCM stereo (mono is stereo is not supported) with the current sample rate |
| int defBits = 16; |
| int defChannels = MIN(2, channelsCount); |
| float defSampleRate = hardwareSampleRate; |
| // don't add default format is sample rate is not specified |
| bool addDefault = defSampleRate > 0; |
| |
| // TODO: CoreAudio can handle signed/unsigned, little-endian/big-endian |
| // TODO: register the formats (to prevent DirectAudio software conversion) - need to fix DirectAudioDevice.createDataLineInfo |
| // to avoid software conversions if both signed/unsigned or big-/little-endian are supported |
| for (int channelIndex = 0; channelIndex < channelsCount; channelIndex++) { |
| for (int bitIndex = 0; bitIndex < sampleBitsCount; bitIndex++) { |
| int bits = sampleBits[bitIndex]; |
| if (addDefault && bits == defBits && channels[channelIndex] != defChannels && sampleRate == defSampleRate) { |
| // the format is the default one, don't add it now |
| continue; |
| } |
| DAUDIO_AddAudioFormat(creator, |
| bits, // sample size in bits |
| -1, // frame size (auto) |
| channels[channelIndex], // channels |
| sampleRate, // sample rate |
| DAUDIO_PCM, // only accept PCM |
| bits == 8 ? FALSE : TRUE, // signed |
| bits == 8 ? FALSE // little-endian for 8bit |
| : UTIL_IsBigEndianPlatform()); |
| } |
| } |
| // add default format |
| if (addDefault) { |
| DAUDIO_AddAudioFormat(creator, |
| defBits, // 16 bits |
| -1, // automatically calculate frame size |
| defChannels, // channels |
| defSampleRate, // sample rate |
| DAUDIO_PCM, // PCM |
| TRUE, // signed |
| UTIL_IsBigEndianPlatform()); // native endianess |
| } |
| |
| TRACE0("<<DAUDIO_GetFormats\n"); |
| } |
| |
| |
| // ======================================= |
| // Source/Target DataLine functions implementation |
| |
| // ==== |
| /* 1writer-1reader ring buffer class with flush() support */ |
| class RingBuffer { |
| public: |
| RingBuffer() : pBuffer(NULL), nBufferSize(0) { |
| pthread_mutex_init(&lockMutex, NULL); |
| } |
| ~RingBuffer() { |
| Deallocate(); |
| pthread_mutex_destroy(&lockMutex); |
| } |
| |
| // extraBytes: number of additionally allocated bytes to prevent data |
| // overlapping when almost whole buffer is filled |
| // (required only if Write() can override the buffer) |
| bool Allocate(int requestedBufferSize, int extraBytes) { |
| int fullBufferSize = requestedBufferSize + extraBytes; |
| int powerOfTwo = 1; |
| while (powerOfTwo < fullBufferSize) { |
| powerOfTwo <<= 1; |
| } |
| pBuffer = (Byte*)malloc(powerOfTwo); |
| if (pBuffer == NULL) { |
| ERROR0("RingBuffer::Allocate: OUT OF MEMORY\n"); |
| return false; |
| } |
| |
| nBufferSize = requestedBufferSize; |
| nAllocatedBytes = powerOfTwo; |
| nPosMask = powerOfTwo - 1; |
| nWritePos = 0; |
| nReadPos = 0; |
| nFlushPos = -1; |
| |
| TRACE2("RingBuffer::Allocate: OK, bufferSize=%d, allocated:%d\n", nBufferSize, nAllocatedBytes); |
| return true; |
| } |
| |
| void Deallocate() { |
| if (pBuffer) { |
| free(pBuffer); |
| pBuffer = NULL; |
| nBufferSize = 0; |
| } |
| } |
| |
| inline int GetBufferSize() { |
| return nBufferSize; |
| } |
| |
| inline int GetAllocatedSize() { |
| return nAllocatedBytes; |
| } |
| |
| // gets number of bytes available for reading |
| int GetValidByteCount() { |
| lock(); |
| INT64 result = nWritePos - (nFlushPos >= 0 ? nFlushPos : nReadPos); |
| unlock(); |
| return result > (INT64)nBufferSize ? nBufferSize : (int)result; |
| } |
| |
| int Write(void *srcBuffer, int len, bool preventOverflow) { |
| lock(); |
| TRACE2("RingBuffer::Write (%d bytes, preventOverflow=%d)\n", len, preventOverflow ? 1 : 0); |
| TRACE2(" writePos = %lld (%d)", (long long)nWritePos, Pos2Offset(nWritePos)); |
| TRACE2(" readPos=%lld (%d)", (long long)nReadPos, Pos2Offset(nReadPos)); |
| TRACE2(" flushPos=%lld (%d)\n", (long long)nFlushPos, Pos2Offset(nFlushPos)); |
| |
| INT64 writePos = nWritePos; |
| if (preventOverflow) { |
| INT64 avail_read = writePos - (nFlushPos >= 0 ? nFlushPos : nReadPos); |
| if (avail_read >= (INT64)nBufferSize) { |
| // no space |
| TRACE0(" preventOverlow: OVERFLOW => len = 0;\n"); |
| len = 0; |
| } else { |
| int avail_write = nBufferSize - (int)avail_read; |
| if (len > avail_write) { |
| TRACE2(" preventOverlow: desrease len: %d => %d\n", len, avail_write); |
| len = avail_write; |
| } |
| } |
| } |
| unlock(); |
| |
| if (len > 0) { |
| |
| write((Byte *)srcBuffer, Pos2Offset(writePos), len); |
| |
| lock(); |
| TRACE4("--RingBuffer::Write writePos: %lld (%d) => %lld, (%d)\n", |
| (long long)nWritePos, Pos2Offset(nWritePos), (long long)nWritePos + len, Pos2Offset(nWritePos + len)); |
| nWritePos += len; |
| unlock(); |
| } |
| return len; |
| } |
| |
| int Read(void *dstBuffer, int len) { |
| lock(); |
| TRACE1("RingBuffer::Read (%d bytes)\n", len); |
| TRACE2(" writePos = %lld (%d)", (long long)nWritePos, Pos2Offset(nWritePos)); |
| TRACE2(" readPos=%lld (%d)", (long long)nReadPos, Pos2Offset(nReadPos)); |
| TRACE2(" flushPos=%lld (%d)\n", (long long)nFlushPos, Pos2Offset(nFlushPos)); |
| |
| applyFlush(); |
| INT64 avail_read = nWritePos - nReadPos; |
| // check for overflow |
| if (avail_read > (INT64)nBufferSize) { |
| nReadPos = nWritePos - nBufferSize; |
| avail_read = nBufferSize; |
| TRACE0(" OVERFLOW\n"); |
| } |
| INT64 readPos = nReadPos; |
| unlock(); |
| |
| if (len > (int)avail_read) { |
| TRACE2(" RingBuffer::Read - don't have enough data, len: %d => %d\n", len, (int)avail_read); |
| len = (int)avail_read; |
| } |
| |
| if (len > 0) { |
| |
| read((Byte *)dstBuffer, Pos2Offset(readPos), len); |
| |
| lock(); |
| if (applyFlush()) { |
| // just got flush(), results became obsolete |
| TRACE0("--RingBuffer::Read, got Flush, return 0\n"); |
| len = 0; |
| } else { |
| TRACE4("--RingBuffer::Read readPos: %lld (%d) => %lld (%d)\n", |
| (long long)nReadPos, Pos2Offset(nReadPos), (long long)nReadPos + len, Pos2Offset(nReadPos + len)); |
| nReadPos += len; |
| } |
| unlock(); |
| } else { |
| // underrun! |
| } |
| return len; |
| } |
| |
| // returns number of the flushed bytes |
| int Flush() { |
| lock(); |
| INT64 flushedBytes = nWritePos - (nFlushPos >= 0 ? nFlushPos : nReadPos); |
| nFlushPos = nWritePos; |
| unlock(); |
| return flushedBytes > (INT64)nBufferSize ? nBufferSize : (int)flushedBytes; |
| } |
| |
| private: |
| Byte *pBuffer; |
| int nBufferSize; |
| int nAllocatedBytes; |
| INT64 nPosMask; |
| |
| pthread_mutex_t lockMutex; |
| |
| volatile INT64 nWritePos; |
| volatile INT64 nReadPos; |
| // Flush() sets nFlushPos value to nWritePos; |
| // next Read() sets nReadPos to nFlushPos and resests nFlushPos to -1 |
| volatile INT64 nFlushPos; |
| |
| inline void lock() { |
| pthread_mutex_lock(&lockMutex); |
| } |
| inline void unlock() { |
| pthread_mutex_unlock(&lockMutex); |
| } |
| |
| inline bool applyFlush() { |
| if (nFlushPos >= 0) { |
| nReadPos = nFlushPos; |
| nFlushPos = -1; |
| return true; |
| } |
| return false; |
| } |
| |
| inline int Pos2Offset(INT64 pos) { |
| return (int)(pos & nPosMask); |
| } |
| |
| void write(Byte *srcBuffer, int dstOffset, int len) { |
| int dstEndOffset = dstOffset + len; |
| |
| int lenAfterWrap = dstEndOffset - nAllocatedBytes; |
| if (lenAfterWrap > 0) { |
| // dest.buffer does wrap |
| len = nAllocatedBytes - dstOffset; |
| memcpy(pBuffer+dstOffset, srcBuffer, len); |
| memcpy(pBuffer, srcBuffer+len, lenAfterWrap); |
| } else { |
| // dest.buffer does not wrap |
| memcpy(pBuffer+dstOffset, srcBuffer, len); |
| } |
| } |
| |
| void read(Byte *dstBuffer, int srcOffset, int len) { |
| int srcEndOffset = srcOffset + len; |
| |
| int lenAfterWrap = srcEndOffset - nAllocatedBytes; |
| if (lenAfterWrap > 0) { |
| // need to unwrap data |
| len = nAllocatedBytes - srcOffset; |
| memcpy(dstBuffer, pBuffer+srcOffset, len); |
| memcpy(dstBuffer+len, pBuffer, lenAfterWrap); |
| } else { |
| // source buffer is not wrapped |
| memcpy(dstBuffer, pBuffer+srcOffset, len); |
| } |
| } |
| }; |
| |
| |
| class Resampler { |
| private: |
| enum { |
| kResamplerEndOfInputData = 1 // error to interrupt conversion (end of input data) |
| }; |
| public: |
| Resampler() : converter(NULL), outBuffer(NULL) { } |
| ~Resampler() { |
| if (converter != NULL) { |
| AudioConverterDispose(converter); |
| } |
| if (outBuffer != NULL) { |
| free(outBuffer); |
| } |
| } |
| |
| // inFormat & outFormat must be interleaved! |
| bool Init(const AudioStreamBasicDescription *inFormat, const AudioStreamBasicDescription *outFormat, |
| int inputBufferSizeInBytes) |
| { |
| 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::Init: OK\n"); |
| return true; |
| } |
| |
| // returns size of the internal output buffer |
| int GetOutBufferSize() { |
| return outBufferSize; |
| } |
| |
| // process next part of data (writes resampled data to the ringBuffer without overflow check) |
| int Process(void *srcBuffer, int len, RingBuffer *ringBuffer) { |
| int bytesWritten = 0; |
| TRACE2(">>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("<<AudioConverterFillComplexBuffer: got %d packets (%d bytes)\n", |
| (int)packets, (int)abl.mBuffers[0].mDataByteSize); |
| if (packets > 0) { |
| int bytesToWrite = (int)(packets * asbdOut.mBytesPerPacket); |
| bytesWritten += ringBuffer->Write(abl.mBuffers[0].mData, bytesToWrite, false); |
| } |
| |
| // if outputBuffer is small to store all available frames, |
| // we get noErr here. In the case just continue the conversion |
| } while (err == noErr); |
| |
| if (err != kResamplerEndOfInputData) { |
| // unexpected error |
| OS_ERROR0(err, "Resampler::Process (AudioConverterFillComplexBuffer)"); |
| } |
| } |
| TRACE2("<<Resampler::Process: written %d bytes (converted from %d bytes)\n", bytesWritten, len); |
| |
| return bytesWritten; |
| } |
| |
| // resets internal bufferes |
| void Discontinue() { |
| TRACE0(">>Resampler::Discontinue\n"); |
| if (converter != NULL) { |
| AudioConverterReset(converter); |
| } |
| TRACE0("<<Resampler::Discontinue\n"); |
| } |
| |
| private: |
| AudioConverterRef converter; |
| |
| // buffer for output data |
| // note that there is no problem if the buffer is not big enough to store |
| // all converted data - it's only performance issue |
| void *outBuffer; |
| int outBufferSize; |
| |
| AudioStreamBasicDescription asbdIn; |
| AudioStreamBasicDescription asbdOut; |
| |
| struct InputProcData { |
| Resampler *pThis; |
| Byte *data; // data == NULL means we handle Discontinue(false) |
| int dataSize; // == 0 if all data was already provided to the converted of we handle Discontinue(false) |
| }; |
| |
| static OSStatus ConverterInputProc(AudioConverterRef inAudioConverter, UInt32 *ioNumberDataPackets, |
| AudioBufferList *ioData, AudioStreamPacketDescription **outDataPacketDescription, void *inUserData) |
| { |
| InputProcData *data = (InputProcData *)inUserData; |
| |
| TRACE3(" >>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(" <<ConverterInputProc: returns kResamplerEndOfInputData\n"); |
| return kResamplerEndOfInputData; |
| } |
| |
| ioData->mNumberBuffers = 1; |
| ioData->mBuffers[0].mNumberChannels = data->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(" <<ConverterInputProc: returns %d packets\n", (int)(*ioNumberDataPackets)); |
| return noErr; |
| } |
| |
| }; |
| |
| |
| struct OSX_DirectAudioDevice { |
| AudioUnit audioUnit; |
| RingBuffer ringBuffer; |
| AudioStreamBasicDescription asbd; |
| |
| // only for target lines |
| UInt32 inputBufferSizeInBytes; |
| Resampler *resampler; |
| // to detect discontinuity (to reset resampler) |
| SInt64 lastWrittenSampleTime; |
| |
| |
| OSX_DirectAudioDevice() : audioUnit(NULL), asbd(), resampler(NULL), lastWrittenSampleTime(0) { |
| } |
| |
| ~OSX_DirectAudioDevice() { |
| if (audioUnit) { |
| AudioComponentInstanceDispose(audioUnit); |
| } |
| if (resampler) { |
| delete resampler; |
| } |
| } |
| }; |
| |
| static AudioUnit CreateOutputUnit(AudioDeviceID deviceID, int isSource) |
| { |
| OSStatus err; |
| AudioUnit unit; |
| |
| AudioComponentDescription desc; |
| desc.componentType = kAudioUnitType_Output; |
| desc.componentSubType = (deviceID == 0 && isSource) ? kAudioUnitSubType_DefaultOutput : kAudioUnitSubType_HALOutput; |
| desc.componentManufacturer = kAudioUnitManufacturer_Apple; |
| desc.componentFlags = 0; |
| desc.componentFlagsMask = 0; |
| |
| AudioComponent comp = AudioComponentFindNext(NULL, &desc); |
| err = AudioComponentInstanceNew(comp, &unit); |
| |
| if (err) { |
| OS_ERROR0(err, "CreateOutputUnit:OpenAComponent"); |
| return NULL; |
| } |
| |
| if (!isSource) { |
| int enableIO = 0; |
| err = AudioUnitSetProperty(unit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, |
| 0, &enableIO, sizeof(enableIO)); |
| if (err) { |
| OS_ERROR0(err, "SetProperty (output EnableIO)"); |
| } |
| enableIO = 1; |
| err = AudioUnitSetProperty(unit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, |
| 1, &enableIO, sizeof(enableIO)); |
| if (err) { |
| OS_ERROR0(err, "SetProperty (input EnableIO)"); |
| } |
| |
| if (!deviceID) { |
| // get real AudioDeviceID for default input device (macosx current input device) |
| deviceID = GetDefaultDevice(isSource); |
| if (!deviceID) { |
| AudioComponentInstanceDispose(unit); |
| return NULL; |
| } |
| } |
| } |
| |
| if (deviceID) { |
| err = AudioUnitSetProperty(unit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, |
| 0, &deviceID, sizeof(deviceID)); |
| if (err) { |
| OS_ERROR0(err, "SetProperty (CurrentDevice)"); |
| AudioComponentInstanceDispose(unit); |
| return NULL; |
| } |
| } |
| |
| return unit; |
| } |
| |
| static OSStatus OutputCallback(void *inRefCon, |
| AudioUnitRenderActionFlags *ioActionFlags, |
| const AudioTimeStamp *inTimeStamp, |
| UInt32 inBusNumber, |
| UInt32 inNumberFrames, |
| AudioBufferList *ioData) |
| { |
| OSX_DirectAudioDevice *device = (OSX_DirectAudioDevice*)inRefCon; |
| |
| int nchannels = ioData->mNumberBuffers; // should be always == 1 (interleaved channels) |
| AudioBuffer *audioBuffer = ioData->mBuffers; |
| |
| TRACE3(">>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("<<OutputCallback (returns %d)\n", bytesRead); |
| |
| return noErr; |
| } |
| |
| static OSStatus InputCallback(void *inRefCon, |
| AudioUnitRenderActionFlags *ioActionFlags, |
| const AudioTimeStamp *inTimeStamp, |
| UInt32 inBusNumber, |
| UInt32 inNumberFrames, |
| AudioBufferList *ioData) |
| { |
| OSX_DirectAudioDevice *device = (OSX_DirectAudioDevice*)inRefCon; |
| |
| TRACE4(">>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, "<<InputCallback: AudioUnitRender"); |
| } else { |
| if (device->resampler != NULL) { |
| // test for discontinuity |
| // AUHAL starts timestamps at zero, so test if the current timestamp less then the last written |
| SInt64 sampleTime = inTimeStamp->mSampleTime; |
| if (sampleTime < device->lastWrittenSampleTime) { |
| // discontinuity, reset the resampler |
| TRACE2(" InputCallback (RESAMPLED), DISCONTINUITY (%f -> %f)\n", |
| (float)device->lastWrittenSampleTime, (float)sampleTime); |
| |
| device->resampler->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("<<InputCallback (RESAMPLED, saved %d bytes of %d)\n", bytesWritten, (int)abl.mBuffers[0].mDataByteSize); |
| } else { |
| int bytesWritten = device->ringBuffer.Write(abl.mBuffers[0].mData, (int)abl.mBuffers[0].mDataByteSize, false); |
| TRACE2("<<InputCallback (saved %d bytes of %d)\n", bytesWritten, (int)abl.mBuffers[0].mDataByteSize); |
| } |
| } |
| |
| return noErr; |
| } |
| |
| |
| static void FillASBDForNonInterleavedPCM(AudioStreamBasicDescription& asbd, |
| float sampleRate, int channels, int sampleSizeInBits, bool isFloat, int isSigned, bool isBigEndian) |
| { |
| // FillOutASBDForLPCM cannot produce unsigned integer format |
| asbd.mSampleRate = sampleRate; |
| asbd.mFormatID = kAudioFormatLinearPCM; |
| asbd.mFormatFlags = (isFloat ? kAudioFormatFlagIsFloat : (isSigned ? kAudioFormatFlagIsSignedInteger : 0)) |
| | (isBigEndian ? (kAudioFormatFlagIsBigEndian) : 0) |
| | kAudioFormatFlagIsPacked; |
| asbd.mBytesPerPacket = channels * ((sampleSizeInBits + 7) / 8); |
| asbd.mFramesPerPacket = 1; |
| asbd.mBytesPerFrame = asbd.mBytesPerPacket; |
| asbd.mChannelsPerFrame = channels; |
| asbd.mBitsPerChannel = sampleSizeInBits; |
| } |
| |
| void* DAUDIO_Open(INT32 mixerIndex, INT32 deviceID, int isSource, |
| int encoding, float sampleRate, int sampleSizeInBits, |
| int frameSize, int channels, |
| int isSigned, int isBigEndian, int bufferSizeInBytes) |
| { |
| TRACE3(">>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("<<DAUDIO_Open: ERROR: unsupported encoding (%d)\n", encoding); |
| return NULL; |
| } |
| if (channels <= 0) { |
| ERROR1("<<DAUDIO_Open: ERROR: Invalid number of channels=%d!\n", channels); |
| return NULL; |
| } |
| |
| OSX_DirectAudioDevice *device = new OSX_DirectAudioDevice(); |
| |
| AudioUnitScope scope = isSource ? kAudioUnitScope_Input : kAudioUnitScope_Output; |
| int element = isSource ? 0 : 1; |
| OSStatus err = noErr; |
| int extraBufferBytes = 0; |
| |
| device->audioUnit = CreateOutputUnit(deviceID, isSource); |
| |
| if (!device->audioUnit) { |
| delete device; |
| return NULL; |
| } |
| |
| if (!isSource) { |
| AudioDeviceID actualDeviceID = deviceID != 0 ? deviceID : GetDefaultDevice(isSource); |
| float hardwareSampleRate = GetSampleRate(actualDeviceID, isSource); |
| TRACE2("--DAUDIO_Open: sampleRate = %f, hardwareSampleRate=%f\n", sampleRate, hardwareSampleRate); |
| |
| if (fabs(sampleRate - hardwareSampleRate) > 1) { |
| device->resampler = new Resampler(); |
| |
| // request HAL for Float32 with native endianess |
| FillASBDForNonInterleavedPCM(device->asbd, hardwareSampleRate, channels, 32, true, false, kAudioFormatFlagsNativeEndian != 0); |
| } else { |
| sampleRate = hardwareSampleRate; // in case sample rates are not exactly equal |
| } |
| } |
| |
| if (device->resampler == NULL) { |
| // no resampling, request HAL for the requested format |
| FillASBDForNonInterleavedPCM(device->asbd, sampleRate, channels, sampleSizeInBits, false, isSigned, isBigEndian); |
| } |
| |
| err = AudioUnitSetProperty(device->audioUnit, kAudioUnitProperty_StreamFormat, scope, element, &device->asbd, sizeof(device->asbd)); |
| if (err) { |
| OS_ERROR0(err, "<<DAUDIO_Open set StreamFormat"); |
| delete device; |
| return NULL; |
| } |
| |
| AURenderCallbackStruct output; |
| output.inputProc = isSource ? OutputCallback : InputCallback; |
| output.inputProcRefCon = device; |
| |
| err = AudioUnitSetProperty(device->audioUnit, |
| isSource |
| ? (AudioUnitPropertyID)kAudioUnitProperty_SetRenderCallback |
| : (AudioUnitPropertyID)kAudioOutputUnitProperty_SetInputCallback, |
| kAudioUnitScope_Global, 0, &output, sizeof(output)); |
| if (err) { |
| OS_ERROR0(err, "<<DAUDIO_Open set RenderCallback"); |
| delete device; |
| return NULL; |
| } |
| |
| err = AudioUnitInitialize(device->audioUnit); |
| if (err) { |
| OS_ERROR0(err, "<<DAUDIO_Open UnitInitialize"); |
| delete device; |
| return NULL; |
| } |
| |
| if (!isSource) { |
| // for target lines we need extra bytes in the ringBuffer |
| // to prevent collisions when InputCallback overrides data on overflow |
| UInt32 size; |
| OSStatus err; |
| |
| size = sizeof(device->inputBufferSizeInBytes); |
| err = AudioUnitGetProperty(device->audioUnit, kAudioDevicePropertyBufferFrameSize, kAudioUnitScope_Global, |
| 0, &device->inputBufferSizeInBytes, &size); |
| if (err) { |
| OS_ERROR0(err, "<<DAUDIO_Open (TargetDataLine)GetBufferSize\n"); |
| delete device; |
| return NULL; |
| } |
| device->inputBufferSizeInBytes *= device->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("<<DAUDIO_Open: resampler.Init() FAILED.\n"); |
| delete device; |
| return NULL; |
| } |
| // extra bytes in the ringBuffer (extraBufferBytes) should be equal resampler output buffer size |
| extraBufferBytes = device->resampler->GetOutBufferSize(); |
| } |
| |
| if (!device->ringBuffer.Allocate(bufferSizeInBytes, extraBufferBytes)) { |
| ERROR0("<<DAUDIO_Open: Ring buffer allocation error\n"); |
| delete device; |
| return NULL; |
| } |
| |
| TRACE0("<<DAUDIO_Open: OK\n"); |
| return device; |
| } |
| |
| int DAUDIO_Start(void* id, int isSource) { |
| OSX_DirectAudioDevice *device = (OSX_DirectAudioDevice*)id; |
| TRACE0("DAUDIO_Start\n"); |
| |
| OSStatus err = AudioOutputUnitStart(device->audioUnit); |
| |
| if (err != noErr) { |
| OS_ERROR0(err, "DAUDIO_Start"); |
| } |
| |
| return err == noErr ? TRUE : FALSE; |
| } |
| |
| int DAUDIO_Stop(void* id, int isSource) { |
| OSX_DirectAudioDevice *device = (OSX_DirectAudioDevice*)id; |
| TRACE0("DAUDIO_Stop\n"); |
| |
| OSStatus err = AudioOutputUnitStop(device->audioUnit); |
| |
| return err == noErr ? TRUE : FALSE; |
| } |
| |
| void DAUDIO_Close(void* id, int isSource) { |
| OSX_DirectAudioDevice *device = (OSX_DirectAudioDevice*)id; |
| TRACE0("DAUDIO_Close\n"); |
| |
| delete device; |
| } |
| |
| int DAUDIO_Write(void* id, char* data, int byteSize) { |
| OSX_DirectAudioDevice *device = (OSX_DirectAudioDevice*)id; |
| TRACE1(">>DAUDIO_Write: %d bytes to write\n", byteSize); |
| |
| int result = device->ringBuffer.Write(data, byteSize, true); |
| |
| TRACE1("<<DAUDIO_Write: %d bytes written\n", result); |
| return result; |
| } |
| |
| int DAUDIO_Read(void* id, char* data, int byteSize) { |
| OSX_DirectAudioDevice *device = (OSX_DirectAudioDevice*)id; |
| TRACE1(">>DAUDIO_Read: %d bytes to read\n", byteSize); |
| |
| int result = device->ringBuffer.Read(data, byteSize); |
| |
| TRACE1("<<DAUDIO_Read: %d bytes has been read\n", result); |
| return result; |
| } |
| |
| int DAUDIO_GetBufferSize(void* id, int isSource) { |
| OSX_DirectAudioDevice *device = (OSX_DirectAudioDevice*)id; |
| |
| int bufferSizeInBytes = device->ringBuffer.GetBufferSize(); |
| |
| TRACE1("DAUDIO_GetBufferSize returns %d\n", bufferSizeInBytes); |
| return bufferSizeInBytes; |
| } |
| |
| int DAUDIO_StillDraining(void* id, int isSource) { |
| OSX_DirectAudioDevice *device = (OSX_DirectAudioDevice*)id; |
| |
| int draining = device->ringBuffer.GetValidByteCount() > 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 |
